[SR-Users] Kamailio config syntax check leads to memory error
Hi everyone, I'm using Kamailio 3.2.1 with a quite big config file. After adding some new config lines I did a syntax check of the configuration file with "kamailio -c -f " and got the following error output: loading modules under /opt/kamailio/lib/kamailio/modules_k/:/opt/kamailio/lib/kamailio/modules/ 0(28755) : [cfg.lex:1426]: ERROR:lex:addstr: memory allocation error 0(28755) : [cfg.lex:1428]: ERROR:lex:addstr: try to increase pkg size with -M parameter increasing the pkg memory (-M) or shared memory (-m) has no effect. I played around with the size of the configuration file with xlog lines and the cause really seems to be the size of the config file. Restart of Kamailio is possible without problems, it is only the syntax check that leads to the memory error output Is there any possibility to increase the memory for the syntax check? Regards Fred ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Siremis 4.1.0 menu adjustment for different users
Hi, I'm setting up a Registrar server using Kamailio 4.1.3 with Siremis 4.1.0 as WebGUI. On Siremis I struggle to set up different users which only see limited menu items. What I want to do is e.g. an user called "CustomerService" which only sees the "Subscriber_List" menu item and a second user called "SystemAdmin" which only sees the "Dispatcher_List" menu item. I tried different things with new roles, groups and adjustments in the menu administration but didn't manage to get the desired result. Does anyone have a hint for me if that is possible and how? Thanks Fred ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] duplicate information in siptrace table
Hello,
I use Kamailio 4.1.3 with siptrace module. In the sip_trace table I see
duplicate information, one line with empty and one line with filled
traced_user column.
The empty lines are useless for me, I would like only entries with
traced_user information.
Here the siptrace part of kamailio.cfg:
Parameters for Siptrace module
modparam("siptrace", "db_url", DBURL)
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "trace_sl_acks", 0)
modparam("siptrace", "traced_user_avp", "$avp(s:user)")
modparam("siptrace", "trace_delayed", 0)
## SIP Trace Config in main route
if (!is_method("OPTIONS")) {
if (!ds_is_from_list("2")) {
$avp(s:user) = $fU;
} else {
$avp(s:user) = $rU;
};
if (is_avp_set("$avp(s:user)")){
sip_trace();
#setflag(22);
};
};
Has anyone a tip for me how I can get rid of the empty traced_user lines?
Regards
Fred
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Re: [SR-Users] [newbie] questions
For the sake of argument, let's say it was a smart-ass answer-- if your post was to "be pointed in the right direction where I could get (further) help" then that next help should either be: a) the wiki b) a training class c) a consultant On forums I notice that those who get a better response usually use their real name instead of "Me" with a private/alias email address. To breakdown how I view your mail... if you're looking for a proxy, then yes, Kamailio is a great choice. If you're looking for an SBC, then no... Kamailio alone isn't your best choice; although it could be with some tweaking and (possible) use of other products. If you're looking for someone to help you with this complex question, and to learn about this product, I recommend B & C. With best regards, Fred Posner http://qxork.com On Jan 30, 2012, at 7:13 PM, Me wrote: > >> It's not a smart-arse reply; it's sincere, earnest advice. > Really?! Perhaps you could explain to me how exactly is the "you should get a > consultant" comment on a routine set of questions I posted on a mailing list > created for that very purpose - for Kamailio users like myself - anything > other than a smart-arse reply? > > Does it answer any of my queries? No! > Is it helping me in any way, shape or form? No (do you seriously think I > haven't really thought of "getting a consultant" before posting my queries on > this mailing list?)! > Does it provide any insight or fresh ideas on either what I want to achieve > or the difficulties I am facing, given the problems I described earlier? No! > Does it contribute anything to the discussion on this mailing list, apart > from wasting my own time and bandwidth so that I have to enlighten smart > sparks like the previous poster as well as yourself? No! > >> There's only so much of a massive conceptual nexus that people can >> reasonably traverse on a mailing list. For the most part, mailing lists >> exist to answer specific questions, not provide broad, fundamental guidance >> or extensive pedagogical surveys. >> > I didn't ask for "fundamental guidance" or "pedagogical surveys". I asked to > be pointed in the right direction where I could get (further) help. I did not > force anyone to answer, let alone come up with "gems" like the one I > commented on in my previous post. > > I'd say it again though - this thread is for me to seek > answers/advice/help/guidance - if you, the previous poster, or anyone else > for that matter is unwilling or unable to provide one, then just move along - > there is nothing to see here. Smart-arse comments like "get a consultant" > isn't what I am looking for, nor is the reason for starting this thread on > this mailing list. > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [newbie] questions
On Jan 31, 2012, at 6:32 PM, Me wrote: > Hi, Fred, > > >> For the sake of argument, let's say it was a smart-ass answer-- if your post >> was to "be pointed in the right direction where I could get (further) help" >> then that next help should either be: >> >> a) the wiki >> b) a training class >> c) a consultant >> > >> I'd say it again though - this thread is for me to seek >> answers/advice/help/guidance - if you, the previous poster, or anyone else >> for that matter is unwilling or unable to provide one, then just move along >> - there is nothing to see here. Smart-arse comments like "get a consultant" >> isn't what I am looking for, nor is the reason for starting this thread on >> this mailing list. >> > Can you read? > >> On forums I notice that those who get a better response usually use their >> real name instead of "Me" with a private/alias email address. >> > If I tell you that my real name is Josh Oliveira 'Capo' De Souza would that > make you feel any better? Would you like to know my ssn as well? How about my > date of birth? > > If I tell you where I live would that make you go away and prevent you from > posting off-topic drivel on the above thread or ask silly questions you know > you won't get answers to? TA! > > Josh Josh, I'm sorry-- my response wasn't insulting whatsoever. I'm not sure what your problem is, but to me it looks like you're looking for a fight. I erred on the side of thinking you were above that-- my mistake. I bid you a good day. --fred ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio [No Audio]
I am attempting to route local registered users to local registered users without going to the media server. I have a media server [*] for PSTN. But it doesn't support all the CODECS i wan't to use. Signalling seems fine, but i get NO AUDIO on either call leg. I have the RTP captures from the local side making the call and the media packets seem to be going exactly where the SDP dictates. So i'm a little confused. Is this where an RTP Proxy would come in handy? I haven't been able to get the audio to work without going to the media server. If an RTP Proxy is the answer, how much overhead does the proxy add to the Kamailio server? Is it something that you don't run on the same machine and use a distributed environment of RTP proxies on other servers? Or is this something that should be fine and working without a media proxy? I really appreciate any help! Thank you, Fred ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] $200 bounty
On Jun 20, 2012, at 2:36 PM, copycall wrote: > alex, > > [snip] > > a la carte and table d'hote appreciated. > > thank you, > dave > [snip] I know a great bakery that can offer dessert kamailio pricing. Generally, $6/portion starting. With best regards, Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] $200 bounty
On Jun 20, 2012, at 2:51 PM, copycall wrote: > alex, > > you did. > > how do i know what someone will work for, or the going rate unless i ask? > > yes, you have your company on the business list. > > and yes, we both want to maximize our utilty, cough...i mean keep our money > in our pockets. > > is the list designed for alex balashov to get rich? > > there are dozens of list topics everyday, one item that you don't like > because you think someone is stealing food off your table. > > aren't you being a little presumptuous about who kamailio is for? > > why don't you get a pair and give me a price? > > dave > I have 2 cents sir. To make new friends on this list Don't say "get a pair." The list is designed To share knowledge and support Kamailio If the list were here for alex to become rich the list would be fail here's a great, free tip from a fat baker voip geek you don't know: Be nice. For this list, we all chip in our time at no cost to the best we can. For business, there are many choices, great people charging market rates. On the web, or mail. Take a look. It's in a book. Reading Rainbow. Peace. With best regards, Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] $200 bounty
On Jun 20, 2012, at 2:40 PM, Alex Balashov wrote: > On 06/20/2012 02:38 PM, Fred Posner wrote: > >> I know a great bakery that can offer dessert kamailio pricing. >> Generally, $6/portion starting. > > What's the going rate for the Millenium Falcon cake these days? :-) > They start at $250 and increase by size. =) Death Stars are much cheaper. =) With best regards, Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] $200 bounty
On Jun 20, 2012, at 3:19 PM, David wrote: > Do the death stars you are selling have a patch for the small exhaust port > vulnerability (BBY0) ? > > David > > On 2012-06-20 15:12, Fred Posner wrote: >> On Jun 20, 2012, at 2:40 PM, Alex Balashov wrote: >> >>> On 06/20/2012 02:38 PM, Fred Posner wrote: >>> >>>> I know a great bakery that can offer dessert kamailio pricing. >>>> Generally, $6/portion starting. >>> What's the going rate for the Millenium Falcon cake these days? :-) >>> >> They start at $250 and increase by size. =) >> >> Death Stars are much cheaper. =) >> We've patched the vulnerability but can back port on request. =) With best regards, Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Nathelper module, FLT_NATS, FLT_NATB
On Jun 25, 2012, at 8:57 AM, Richard Brady wrote: > Klaus / Daniel > > Thanks again for assistance with this. > > I've tried the solution based on add_contact_alias() and > handle_ruri_alias() and it works perfectly. > > Richard > Do you have an example of the cfg you can share? Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio NAT traversal
Hi Spencer, Is Kamailio also natted? If so, you may have some issues... if not, it should work great. I run a server like this as well... very happy with it. I gave up on kamailio/freeswitch behind nat. Well, didn't give up, just don't have the time to make it work. With best regards, Fred http://qxork.com On Aug 17, 2012, at 11:04 AM, SamyGo wrote: > Hi, > > You kind of sound a little different here. Are you saying that the > REGISTRATIONs will be handled by Freeswitch but store the registration Data > in Kamailio "location" table !? > > Just go through the Kamailio blog by-Miconda or kb.asipto.com specially the > one on integrating the Asterisk Realtime with kamailio, In that articular > configuration file Kamailio Forwards/relays the Registration attempts to the > Media-Server (FreeSwitch in your case). > > That will get your NAT thing handled atleast. Next thing is , if I'm right > about your requirement, saving of Authenticated users in the Kamailio > locations table and I'm not very sure about how to do this. > > > BR > Sammy > > > On Fri, Aug 17, 2012 at 1:59 AM, Spencer Thomason > wrote: > Hello, > I'd like to use Kamailio in from of FreeSWITCH to handle NAT traversal. Is > there a way I can allow freeswitch to handle all auth but store the > registration in usrloc as I need to send an options ping to the endpoints? > > My thought is something like this: > > natted endpoint -> kamailio -> freeswitch > If the response is 200OK, save the registration. > > Thanks, > Spencer ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio NAT traversal
On my set-up, I forward incoming dids to freeswitch for handling, and then forward requests from registered users to another. There's a great tutorial from Daniel/asipto that should be a very good guide: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc With best regards, Fred http://qxork.com On Aug 17, 2012, at 12:29 PM, Spencer Thomason wrote: > Hi Sammy and Fred, > > Basically I'm building a hosted PBX platform using a muti domain FreeSWITCH > setup. Freeswitch and Kamailio are on a public IP. Previously all endpoints > registered to Freeswitch directly which works great. For scalability > purposes, my thought was to use Kamailio in front of Freeswitch to handle the > grunt work of far end NAT traversal and using rtpproxy for media proxy. > Here's where things get a bit complicated. I don't have a way to handle auth > directly from Kamailio in this particular setup as the application's db > schema will not map to what I need to do multi-domain auth in Kamailio. So > I'm blindly forwarding everything not originating from Freeswitch to > Freeswitch and dealing with auth there. > > Since the client endpoints are behind NAT, this gets interesting :-). I need > to ping all NATed endpoints to keep the connection open. I can't just > forward the REGISTER and let Freeswitch ping the devices because it can't get > to them as they opened a pinhole to Kamailio. I also need presence to work, > and all presence is handled by Freeswitch as well. So my thought is this: > > Let Kamailo ping the endpoints. This obviously requires a registration. > Since Freeswitch is the only piece with credentials, save every REGISTER and > remove the ones that fail in a failure route. I check for a not empty $au > before saving and remove the ones that fail. I'm using usrloc in memory only > mode so I'm not sure about the extra load this would create. > > I need to store the REGISTERs in both Kamailio and Freeswitch. I rewrite the > Contact before forwarding to Freeswitch so that it sends INVITEs back to > Kamailio which can get to the endpoints. I can't use Path as I need the > domain in the RURI. > > I forward SUBSCRIBEs to Freeswitch with the Contact rewritten to point to the > registered user in Kamailio. Freeswitch then sends NOTIFYs to Kamailio which > can get to the endpoints. > > I'm completely open to input on how I might improve this setup or flaws in my > logic but it at least works :-) > > Thanks, > Spencer > > > > On Aug 17, 2012, at 8:04 AM, SamyGo wrote: > >> Hi, >> >> You kind of sound a little different here. Are you saying that the >> REGISTRATIONs will be handled by Freeswitch but store the registration Data >> in Kamailio "location" table !? >> >> Just go through the Kamailio blog by-Miconda or kb.asipto.com specially the >> one on integrating the Asterisk Realtime with kamailio, In that articular >> configuration file Kamailio Forwards/relays the Registration attempts to the >> Media-Server (FreeSwitch in your case). >> >> That will get your NAT thing handled atleast. Next thing is , if I'm right >> about your requirement, saving of Authenticated users in the Kamailio >> locations table and I'm not very sure about how to do this. >> >> >> BR >> Sammy >> >> >> On Fri, Aug 17, 2012 at 1:59 AM, Spencer Thomason >> wrote: >> Hello, >> I'd like to use Kamailio in from of FreeSWITCH to handle NAT traversal. Is >> there a way I can allow freeswitch to handle all auth but store the >> registration in usrloc as I need to send an options ping to the endpoints? >> >> My thought is something like this: >> >> natted endpoint -> kamailio -> freeswitch >> If the response is 200OK, save the registration. >> >> Thanks, >> Spencer ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [ot] virtualization systems
Same preference-- especially when call load gets high or there is conferencing / call recording. ---Fred On Aug 28, 2012, at 2:19 AM, Daniel-Constantin Mierla wrote: > Hello, > > just asking to see your experience deploying sip platforms on virtual > systems. So far I was running Kamailio in virtual machines and no problems, > but I insisted that media servers to be on physical machines. Lately is more > pressure from the market to go everything virtual. > > So the question is more about having everything on virtual systems, proxy and > media server, where the media server can deal with transcoding, conference > rooms and IVRs. > > Any strong comments pro or against? > > What is your preferred virtualization system for such deployments? > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [OT] the role of SBCs
When I first got into VoIP, my knowledge was less than stellar. The main decision make and I had believed that if we hired quality (and not cheap) SME we would be given great information and the money spent would pay for itself. We ended up working with a broadsoft system and an acme packet sbc. We were really sold that this would be the creme de la creme-- no nat issues, failover media, security, stability. Crap. Problems galore, especially with residential NAT users. Despite having 2 acme's in a failover, an outage from the main isp resulted in a crippling thundering herd when connection was restored. Immediately, and know with some decent knowledge, I started working with (at the time) openser. We deployed it within 2 weeks. There was no feature lost. In fact, we had only gains. All the NAT problems suddenly went away. We purposely tried to kill the openser with a thundering herd. Couldn't do it. There was a learning curve, but when is there not a learning curve? Honestly, at that time... the savings (which were incredible) wasn't an issue. If it were more than an acme, we would have paid it. We needed something that worked, and the best product we could find was openser. Since then, I've been a strong supporter. With the recent modifications (do we still consider anti_flood recent?), there's really no other choice for me. Yes, it takes programming, customization, and set-up. So does a commercial product. It's life. When I first deployed kamailio, I didn't consider it an SBC. I considered it an SBC replacement. With best regards, Fred http://qxork.com On Aug 31, 2012, at 3:47 AM, Olle E. Johansson wrote: > In most, but not all, cases it's a political/business decision outside of the > scope of the technichal specifications. A commercial SBC delivers a cloud of > magic dust that makes some people feel better and more secure. I have audited > several SBC installations that are totally insecure, where the local techies > lack knowledge on how to operate it. Management people think the SBC is > secure by design. I can't blame the vendors here - it's more correct to blame > the decision process. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] LUA Authentication.
Hi David, I believe this is the example you're looking for. It's on the Asipto KB site: http://kb.asipto.com/kamailio:usage:k32-lua-routing ---fred http://qxork.com On Sep 3, 2012, at 5:06 PM, David | StyleFlare wrote: > I think I saw once an example from miconda using LUA for Auth? > > Does anyone else remember seeing that? > > I wanted to do a custom Auth using LUA. > > Thanks in advance for any pointers. > > David. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio not responding
Do you have an example of the sip traffic as seen by the server? Have you verified kamailio is running with kamctl monitor? ---fred -- Fred Posner http://qxork.com On Nov 6, 2012, at 4:39 PM, wrote: > I have recently installed kamailio on CentOS 6.3 and configured a couple of > SIP phone Linksys and Grandstream to test inercommunication. The SIP-Server/ > Kamailio server ha started and is listen on port 5060. The UA are sending > registered message to the SIP Server but there is no response from the SIP > server. I left the syslog at default but not seeing any messages on the > /var/log/messages. Can someone tell me if I have missed some configuration > information. Note: I registered two users 101 and 202 using the command > kamctl add 101 101. Is it possible to check whether the Users added have been > added to the system??. and does anyone have any idea how to to troubleshoot a > response from the Server. I am using Wireshark to monitor communication > between between UA and SIP Server > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SIP Ping utility with kamailio
Hey JR...
I use this:
#! /usr/bin/perl -w
use IO::Socket;
use POSIX 'strftime';
my ($msg,$remotehost,$callid,$socket,$date,$branch,$localip,$dest);
$remotehost = $ARGV[0]
or die "FAIL \(no host defined\)\n";
if ($ARGV[1]) {
$remoteport = $ARGV[1];
} else {
$remoteport = "5060";
}
if ($ARGV[2]) {
$localip = $ARGV[2];
} else {
$localip = "127.0.0.1";
}
if ($ARGV[2]) {
$dest = $ARGV[3];
} else {
$dest = "ping";
}
$socket = IO::Socket::INET->new (
PeerAddr => $remotehost,
PeerPort => $remoteport,
Proto => 'udp',
) or die "FAIL Could not create socket: $!n";
$callid .= ('0'..'9', "a".."f")[int(rand(16))] for 1 .. 32;
$date = strftime('%a, %e %B %Y %I:%M:%S %Z',localtime());
$branch="z9hG4bk" . time();
my $packet = qq(OPTIONS sip:$remotehost SIP/2.0
Via: SIP/2.0/UDP $localip:$remoteport;branch=$branch
From:
To:
Contact: http://qxork.com
On Oct 15, 2010, at 11:14 AM, JR Richardson wrote:
> Hi All,
>
> Can someone point me in the right direction of a command line SIP Ping
> utility or how to invoke from Kamailio? I see there is a sip_ping.pl
> script in voip-hacks, does anyone have copy-paste text version of
> that, all I can find is the PDF?
>
> Thanks.
>
> JR
> --
> JR Richardson
> Engineering for the Masses
>
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Re: [SR-Users] SIP Ping utility with kamailio
On Oct 15, 2010, at 11:42 AM, JR Richardson wrote:
> On Fri, Oct 15, 2010 at 10:22 AM, Fred Posner wrote:
>> Hey JR...
>>
>> I use this:
>>
>> #! /usr/bin/perl -w
>> use IO::Socket;
>> use POSIX 'strftime';
>>
>> my ($msg,$remotehost,$callid,$socket,$date,$branch,$localip,$dest);
>>
>> $remotehost = $ARGV[0]
>>or die "FAIL \(no host defined\)\n";
>>
>> if ($ARGV[1]) {
>>$remoteport = $ARGV[1];
>>} else {
>>$remoteport = "5060";
>> }
>>
>> if ($ARGV[2]) {
>>$localip = $ARGV[2];
>>} else {
>>$localip = "127.0.0.1";
>> }
>>
>> if ($ARGV[2]) {
>>$dest = $ARGV[3];
>>} else {
>>$dest = "ping";
>> }
>>
>> $socket = IO::Socket::INET->new (
>>PeerAddr => $remotehost,
>>PeerPort => $remoteport,
>>Proto => 'udp',
>>) or die "FAIL Could not create socket: $!n";
>>
>> $callid .= ('0'..'9', "a".."f")[int(rand(16))] for 1 .. 32;
>> $date = strftime('%a, %e %B %Y %I:%M:%S %Z',localtime());
>> $branch="z9hG4bk" . time();
>>
>> my $packet = qq(OPTIONS sip:$remotehost SIP/2.0
>> Via: SIP/2.0/UDP $localip:$remoteport;branch=$branch
>> From:
>> To:
>> Contact: > Call-ID: $call...@$localip
>> CSeq: 102 Options
>> User-Agent: sipcheck.pl
>> Date: $date
>> Allow: ACK, CANCEL
>> Content-Length: 0
>> );
>>
>>
>> print $socket $packet;
>>
>> eval {
>>local $SIG{ALRM} = sub { die };
>>alarm 5;
>>my $sock_addr = recv($socket,$msg,190,0);
>>alarm 0;
>>1;
>> } or die("FAIL\n");
>>
>> if ($msg) {
>>print "UP\n";
>>print "response is $msg\n";
>> } else {
>>print "FAIL no msg received\n";
>> }
>> close($socket);
>>
>>
>> ---fred
>> http://qxork.com
>
> Thanks Fred, this works fine, I like the response, you get more info
> than the 'kamctl ping".
>
> I'm looking more for a utility that I can run in the background and
> graph latency, that gives me specific time value so I can dump into
> MRTG.
>
> Any ideas?
>
> JR
>
> --
> JR Richardson
> Engineering for the Masses
Give this one a try... I think it's the text version of the pdf you were hoping
for:
http://flylib.com/books/en/3.439.1.97/1/
---fred
http://qxork.com
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Re: [SR-Users] Creating multiple Accounts on kamailio with script.
On Nov 15, 2010, at 5:28 PM, Siddhardha Garige wrote: > Hello all, > > I am conducting some load testing and need to create 50,000 accounts on > kamailio. > > I want to write a shell script to add 50,000 users. Kamctl add [user] > [password] prompts for a password and I checked kamctl -help and there is no > option for providing password along with the command. > > Is is possible to disable password and run my script? > > Thanks > Sid Why not just insert into the DB directly? ---fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SIP Scanning Attacks Experiences
On Nov 18, 2010, at 8:49 AM, marius zbihlei wrote: > On 11/18/2010 01:58 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> during the testing period of Kamailio 3.1.0, while running it at >> voipuser.org, I had the chance to watch live and analyze a SIP scanning >> attack. Yesterday I noticed another one by looking at Siremis 2.0 >> charts, therefore I wrote an article with some hints about what you can >> use to protect your SIP services within Kamailio configuration file. >> >> You can read it at: >>* http://asipto.com/u/i >> >> Hope is going to be useful for many of you! >> >> Cheers, >> Daniel >> >> > Hello Daniel, > > Nice read, thanks for sharing. This "friendly-scanner" messages has really > gotten out of hand lately. FYI, they are generated by a python suite called > SIPVicious (ha ha nice pun)(http://code.google.com/p/sipvicious/) . More on > this http://blog.sipvicious.org/. The suite was developed (really really > extended the sense of the word "developed" here - as the scripts are really > basic) by a security company who trails over Europe giving lectures on Voip > security. :) > > Cheers, > Marius SIP Vicious does have a kill command... I've tried launching that on detection with mixed results. Triggering it from a hash count might prove better. With best regards, Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SBC on kamailio
On Mon, 2010-12-20 at 17:00 +0100, Henry Dogger wrote: > Hi all, > > We have a kamailio installation for our voip system and would like to > add a SBC. > > We found the freeswitch possibility: > http://wiki.freeswitch.org/wiki/SBC_Setup > > But in this case, 302 Redirect SIP is used, we would like the SBC to > act as a pass-through controller, just to be a gateway for SIP clients > from outside. > > Is something like this possible? And will registrations be handled > correctly since all SIP messages will be on port 5060. > > Thanks in advance. > > Kind regards, > Henry Dogger > Telecats BV There's a great tutorial for Kamailio / Freeswitch on the Asipto site: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc -- With best regards, Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SBC on kamailio
On Mon, 2010-12-20 at 17:26 +0100, Henry Dogger wrote: > Thanks, but we have a different setup... > We have clients (phones) connecting from the internet, and will register and > call to externalIP:5060. > Is a setup like this possible as well? > > Henry > > -Original Message- > On Mon, 2010-12-20 at 17:00 +0100, Henry Dogger wrote: > > Hi all, > > > > We have a kamailio installation for our voip system and would like to > > add a SBC. > > > > We found the freeswitch possibility: > > http://wiki.freeswitch.org/wiki/SBC_Setup > > > > But in this case, 302 Redirect SIP is used, we would like the SBC to > > act as a pass-through controller, just to be a gateway for SIP clients > > from outside. > > > > Is something like this possible? And will registrations be handled > > correctly since all SIP messages will be on port 5060. > > > > Thanks in advance. > > > > Kind regards, > > Henry Dogger > > Telecats BV > > There's a great tutorial for Kamailio / Freeswitch on the Asipto site: > > http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc > Yes. That's pretty straight forward in the tutorial. -- With best regards, Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SBC on kamailio
(not top posting) On Tue, 2010-12-21 at 15:14 +0100, Henry Dogger wrote: > Any ideas for setting up a setup like this? > > -Original Message- > From: sr-users-boun...@lists.sip-router.org > [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Henry Dogger > Sent: dinsdag 21 december 2010 10:50 > To: Klaus Darilion > Cc: sr-users@lists.sip-router.org > Subject: Re: [SR-Users] SBC on kamailio > > We would like to use the SBC, between customers and Kamailio. > Since we have a few systems behind our kamailio where customers are > routed with reverse routing, we want to shield our internal IP > addresses. > It sounds like you really just want topology hiding... you can use the topoh module for that. http://kamailio.org/docs/modules/3.1.x/modules/topoh.html http://by-miconda.blogspot.com/2010/01/best-of-new-in-kamailio-300-10-topology.html -- With best regards, Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Support for via ;maddr
On Tue, 2011-01-04 at 18:24 +0100, Olle E. Johansson wrote: > Friends, > We've had an interesting discussion on the Asterisk-dev mailing list about > supporting the ;maddr and ;ttl attributes in the via header when sending > responses. We've agreed that it should be considered harmful and suggest > making it configurable whether to support it in Asterisk. > > My question now is > > - Does kamailio support this automatically? > - Can I disable it? > > Regards, > /O I believe it's with trasformations the pv (psuedo variables) module. http://www.kamailio.org/dokuwiki/doku.php/transformations:3.1.x http://www.kamailio.org/docs/modules/3.1.x/modules_k/pv.html Isn't it something that is required to be looked at via RFC3261? -- With best regards, Fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio and FreeSWITCH realtime integration, tutorial?
It's right on the site: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms With best regards, Fred http://qxork.com On Sep 21, 2011, at 3:46 PM, Henrik Aagaard Sørensen wrote: > Does anyone know if there somewhere exists a tutorial about Kamailio and > FreeSWITCH realtime integration? > > I have Googled a lot and found: > http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index > > On the same site there is a tutorial for Kamailio and Asterisk realtime > integration, but not for FreeSWITCH. > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Dropped registrar bindings with 1.5.3
On Nov 18, 2011, at 5:47 AM, Klaus Darilion wrote:
>
>
> On 18.11.2011 05:07, Alex Balashov wrote:
>> On 11/17/2011 11:02 PM, Juha Heinanen wrote:
>>
>>>> else if(is_method("REGISTER")) {
>>>> xlog("L_INFO", "... Processing REGISTER from $si:$sp for
>>>> AOR $tu\n");
>>>> route(2);
>>>> exit;
>>>> }
>>>
What happens if you move the Register to the beginning of the if statement?
With best regards,
Fred
http://qxork.com
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Re: [SR-Users] Evil SBC/B2BUA, was Re: uac_replace_from recovery with modified header
On 1/25/13 8:54 AM, Alex Balashov wrote: On 01/25/2013 08:51 AM, Daniel-Constantin Mierla wrote: No sense for more, back to testing 4.0.0... Maybe so, but I for one thought it was quite an interesting discussion. :-) I'm more interested in the refund idea. I think I finally found the best way I can contribute to the project; refund administrator. -- fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] #!KAMAILIO
On 1/30/13 5:08 AM, Olle E. Johansson wrote: Daniel, The #!KAMAILIO is something I don't see in documentation. There's some old communication on the various flavour markers in the config file, but can you describe it a bit so we can add it to the core cookbook in the wiki? You referred to it here as well as in the discussion with Philipper earlier. How does it affect variables? /O http://www.kamailio.org/dokuwiki/doku.php/features:new-in-3.0.x#script_compatibility_mode Script compatibility mode option to offer best compatibility with expected behaviour had by older versions as now we integrate functionalities from Kamailio and SER all together, this directive offer the possibility to choose between. For example: in Kamailio, processing of failure route uses selected reply code from branches of the last step of serial forking in SER, processing of failure route uses selected reply code from all branches, offering the option to drop replies for branches of the last step of serial forking via a module function you can choose between ser compatible, kamailio compatible and max compatibility (compatible with both as much as possible), using: #!SER #!KAMAILIO #!OPENSER #!ALL #!MAXCOMPAT where #!KAMAILIO is equivalent with #!OPENSER and #!ALL with #!MAXCOMPAT IMPORTANT - set #!KAMAILIO as first line in your config file if you update from 1.5.x or older version of Kamailio (OpenSER), otherwise you may experience new behaviour for some old functions. #!KAMAILIO ... I've been meaning to check my wiki account, so I'll add this more places. -- fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SIPit 30 summary
On 2/22/13 1:36 PM, Olle E. Johansson wrote: Hi! [snip] Special thank you to Fred Posner who contributed a Big Fred Cookie! One day, I will gladly accept the position of Kamailio Baker, or assist with the cookie cookbook. =) -- fred http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] INVITE messages not authenticated (default configuration)?
I think there's two ways of looking at this...
1) That Kamailio is sending all the calls to Asterisk.
2) That the Asterisk is sending the calls through
I think the post Barry placed on Asterisk list identifies a serious
issue; that being said the easy way one #1 to help avoid this, IMO,...
A) Set a flag after consume credentials
B) Update logic so that any call not intended for a local destination on
that Asterisk box (DID, extension) is then checked for the flag set in
A. If flag isn't there, reject call with 403 or something you wish.
If you have a lot of DIDs, you can do a look up in the routing.
--
fred
http://qxork.com
On 3/8/13 12:00 PM, Barry Flanagan wrote:
On 7 March 2013 22:20, Paul Belanger mailto:paul.belan...@polybeacon.com>> wrote:
Greeting,
Hopefully, I'm understanding the following default kamailio.cfg[1]
file. Over the weekend, I was attached by SipVicious. Following
along with the example Daniel[2] create with kamailio and asterisk, I
have almost the same setup. Rather then storing my SIP profiles in
Asterisk database, I have then in Kamailio.
I also have a test installation originally based on Daniel's example and
have come across the same issue. I also placed a stanza such as the one
below into my [AUTH] route so that INVITES must be authenticated. Given
that in this setup Asterisk is trusting any INVITES from Kamailio it
seems like it should be there for sure.
However, I also found another issue on the Asterisk side related to
this. I raised it on the Asterisk-users list but did not get any
replies. Might be worth a read, and if anyone else here has any idea I
would be grateful. Post is at
http://lists.digium.com/pipermail/asterisk-users/2013-February/277633.html
Regards,
-Barry
To my point, the attacker was actually able to by pass any sort of
authentication, but simply sending an INIVTE message:
./svmap.py -e 18885551234 kamailio.example.org
<http://kamailio.example.org> -m INVITE
Which kamailio, forwarded to Asterisk and because there is no
additional auth within asterisk, was able to hit the asterisk context
for getting processed (they did not get out to the real world).
However, my question is why do we not authenticate INVITE
messages? If my understanding is correct, if would require something
like the following:
if (is_method("INVITE")) {
if (!proxy_authorize("$fd", "subscriber")) {
proxy_challenge("$fd", "0");
exit;
}
}
If so, why not also do it in the default configuration file?
[1]
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=etc/kamailio.cfg;hb=HEAD
[2]
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
--
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Jabber: paul.belan...@polybeacon.com
<mailto:paul.belan...@polybeacon.com> | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
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Re: [SR-Users] home pbx server experience
Do you have an example of the home pbx config you like in kamailio?
Fred Posner | LOD / Team Forrest
ph. 503-914-0999 | f...@lod.com | qxork.com
On 5/14/13 10:27 AM, u wrote:
I would like to share my experience with kamailio and other home pbx servers.
Kamailio on my kirkwood home router for my 6 SIP users is perhaps
overkill: I don't really need mysql and "scalability". But at last I
finally managed to make calling between registered users work stable.
My voip clients only work in all NAT scenarios if I work around some
bugs: to use csipsimple on android I had to change rtpproxy_manage()
to rtpproxy_manage("c") in kamailio's default config, so that problems
with conflicting c: entries in the SDP go away.
I propose kamailio could ship with a special example
kamailio-compatible.cfg that doesn't try to be RFC compliant, but
compatible to the most common voip clients. Right now the only thing I
would change for this is the option for rtpproxy_manage, but I'm sure
others will know more common quirks that could safely be enabled to
increase compatibility. I think this compatibility idea is what yate
sticks to for their defaults. In freeswitch you also have to do it all
manually, and it's much more work to figure things out in their
enormous config files.
The other SIP proxies I had tried before kamailio officially fit all
my requirements, including support for multihomed dynamic IPs, but
contrary to their claims it didn't work.
Yate was easy to set up, but the default dialplan is more confusing
than powerful and after having made everything work I realised yate
was clogging my CPU and RAM and after some time always randomly
stopped working. This is with only 2 users connected! It also wasn't
possible to fix NAT sdp while leaving the codecs section in the SDP
alone at the same time. I tried to debug the code, but the C++ was so
complex that I had to give up.
Freeswitch was much more difficult to setup, a multihomed setup with
dynamic IP was super buggy and it also didn't help that the
unintuitive configuration is all in complex unreadable XML
configuration files.
Kamailio and rtpproxy don't officially support dynamic IP address, but
I can just restart both each time my DSL provider forces me to a new
IP address. This happens automatically in the night and is no big
hassle really. The most simple, least-featureful solution works best
it seems.
Now the last problem I have with kamailio: I don't know how to connect
my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont).
I would like a simple way to do this, preferably without other
features that always seem to complicate the matters. Is there
something more lightweight and simple than asterisk, freeswitch and
yate, that people use successfully for this task together with
kamailio and rtpproxy?
u
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Re: [SR-Users] Issue with RTP proxy....
This is awesome... did it say which version of RTP Proxy or did I just not RTFM well enough? Fred Posner | Team Forrest / LOD direct: 503-914-0999 | fax: 954-472-2896 On 07/02/2013 10:17 AM, Daniel-Constantin Mierla wrote: Hello, have you re-installed rtpproxy from sources after applying the patch I provided the link to? Cheers, Daniel On 7/2/13 3:33 PM, arun Jayaprakash wrote: Hi Daniel, I get this error: rtpproxy -F -l 10.164.62.166 -A 54.x.x.x -s udp:127.0.0.1:7722 rtpproxy: invalid option -- 'A' usage: rtpproxy [-2fvFiPa] [-l addr1[/addr2]] [-6 addr1[/addr2]] [-s path] [-t tos] [-r rdir [-S sdir]] [-T ttl] [-L nfiles] [-m port_min] [-M port_max] [-u uname[:gname]] [-n timeout_socket] [-d log_level[:log_facility]] *From:* Daniel-Constantin Mierla *To:* arun Jayaprakash ; Kamailio (SER) - Users Mailing List *Sent:* Tuesday, July 2, 2013 2:39 AM *Subject:* Re: [SR-Users] Issue with RTP proxy Hello, On 7/1/13 5:48 PM, arun Jayaprakash wrote: Hello, I have enabled RTP proxy on my machine (ec2 instance) by running the following script: rtpproxy -F -l mypublicip -s udp:localhost:7722 but I keep getting this error. The phones ring but there is no audio: 3(4144) ERROR: rtpproxy [rtpproxy.c:2647]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy can someone let me know what this error means? you cannot get rtpproxy listening on the ec2 public ip, which does not exist on the network interface. You will have to use a patched rtpproxy, that adds advertising address support. I had such patch for my own usage, now I polished it a bit and push it to github: - https://github.com/miconda/rtpproxy/commit/41f6d9d9084a6fad52a6483a0593d4b25e0de8ca Give it a try, hope I haven't broken it with last polishing, if does not work let me know. You should be able to apply it to latest rtpproxy stable release. Once you compile and install, then run rtpproxy with: rtpproxy -F -l localip -A publicip -s udp:localhost:7722 Cheers, Daniel -- Daniel-Constantin Mierla -http://www.asipto.com <http://www.asipto.com/> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [OT] Video processing of Kamailio Word recordings
I have a vimeo account that can host a few of them once they are ready. --fred On 8/26/13 12:53 PM, Daniel-Constantin Mierla wrote: Hello, I have few GB with the recordings of presentations from Kamailio World, but they are in full HD format and need to be processed for uploading to one of the web sites hosting videos. Not being familiar with video processing, I am asking here to see if anyone can give some hits about what I could use to prepare the videos for upload. I would need to do: - merge some video files - cut from video files - eventually add few frames at the beginning with title of presentation/name of presenter - resize/transcode to right format for web publishing Let me know if you know some good/easy to use toolkit for such operations (free or at low cost), running on Linux or Mac OS X. Also, if anyone volunteers to help and has already an adequate infrastructure for video editing, maybe it is easier to transfer the raw content via snail mail on a usb stick (unless you are in Berlin area where we can meet in person). Another point would be suggestions about what web sites to use for publishing the videos. Youtube is one of them, but maybe there are better alternatives. If that matters, the average time of presentations should be 25-30min. Thanks, Daniel ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] New Developer: Charles Chance
Welcome Charles! Fred Posner | Team Forrest / LOD direct: 503-914-0999 | fax: 954-472-2896 On 09/13/2013 10:06 AM, Daniel-Constantin Mierla wrote: Hello, I want to announce that a new person got developer GIT write access to repository: Charles Chance. He is for long time in the community, sending patches in the past to modules such as memcache. His immediate goal is to care of dmq module, having a set of patches to be committed as well as plans to integrate it within more modules. His git commit id is: cchance My warm welcome and looking forward to future work within the project! Cheers, Daniel ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] ZRTP
Hi Jonathan, On 10/11/13 3:42 PM, Jonathan Brown wrote: Hi, Does Kamailio support ZRTP end point to end point encryption? If so how is this configured? Sincerely, *Jonathan Brown* Kamailio handles the sip processing of the call allowing the two endpoints to negotiate the zrtp. With Kamailio, you wouldn't need to configure anything special to allow the clients to use zrtp. If both clients support the option (such as jitsi), they can use it to secure the media. -- Fred Posner | The Palner Group, Inc. http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] (no subject)
Hello Sebastien, On 10/11/13 2:10 PM, Sébastien Cramatte wrote: Hello, ... My question is hwo can I replace rtpproxy by ngcp-mediaproxy-ng in bridge mode ? Does it possible ? I use "rwie" and "rwei" flags but in ngcp-mediaproxy-ng e and i seems to be used for IPv4 / IPv6 .. ... I don't believe that mediaproxy-ng can be used to bridge two ipv4 networks; only bridging for ipv6 <-> ipv4. -- Fred Posner http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamilio and AWS Route 53 latency regions
On Mon, 2013-10-14 at 17:12 -0400, Coy Cardwell wrote: > ...Two servers, both sharing the same database. Server1, Server2. Both servers > are behind a NAT (AWS). > > If I set the DNS to return both IP addresses for the domain on the A > record, everything works, all is well. > > If I set the DNS to return values based on latency and the Calling Client > gets Server1's IP Address but the Receiving Client is registered to > Server2, the call fails from a timeout. I can see the attempt to go through > at the packet level and a 'non-local' socket message as well. > > If the DNS returns Server2's IP Address and the Receiving Client is > registered to Server2, the call completes... > > Thanks. > > - Coy Cardwell Hi Coy, Can you explain the scenario a bit in regards to sharing the same database across the two servers? I can be a problem for the local client to try to receive a call from a server that they're not registered to. --fred Fred Posner, @qxork ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio failed to start
It appears as though you might have multiple syntax errors in the script -- such as not opening or not closing certain tags. The best way to troubleshoot this (or at least the way I like best) is to go back to your last known good config and make the changes one at a time. --fred Wingsravi R wrote: >___ >SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >sr-users@lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio network edge for registration and rtp pass-through
Do you need the registration be local to the asterisk? I would have all the asterisks send calls to the Kamailio. You can have a lookup on endpoint outbound to decide which asterisk should handle the outbound call for that did. Also a lookup for incoming DIDs, etc. ---Fred > On Oct 25, 2013, at 5:43 PM, Jr Richardson wrote: > > Hi All, > > Starting a new project, roll your own SBC, not a full SBC, just need some > minor functionality. I'm interested in deploying Kamailio as a edge device > on a VSP for single entry point for hosted PBX's, Asterisk based. I had some > wonderful and informative conversations at Astricon 2013, several folks > assuring me Kamailio w/rtpproxy was the tool for the job, so this is a follow > up to delve more into details. > > I've been researching configs, topology, modules needed, ect... Most of the > examples I'm reading about for this scenario are spreading the registrations > across many PBX's without distinction. One concept I'm struggling with is > having a specific phone register to a specific PBX. > > phone-customer-A-x101> phone-customer-B-x101> phone-customer-C-x101> > I could add a unique identifier to some part of the registration of each > phone like 'custA-101@kamailio_server', custB-101@kamailio_server, ect. What > I'm not clear on is when the request comes to kamailio, where would I > identify what PBX the phone should register to and how to re-write the > 'custA-101@kamailio_server' to '101@custA-pbx' and forward to the correct PBX > and ensure rtp flows through kamailio. > > Could this function be derived using dbaliases or possibly using dispatcher > with group number for each customer PBX? > > So assuming I can get the registrations to work properly, would standard > invite for calling just work or would I also have to have specific config in > place to ensure an invite from customer A phone also reaches the correct > customer A PBX? > > A point in the right direction? > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] TLS - No ServerHello
There's a great debugging article posted to: http://www.kamailio.org/wiki/tutorials/tls/testing-and-debugging What kind of response do you get from: openssl s_client -connect IPADDRESS:5061 -no_ssl2 -bugs Fred Posner | The Palner Group direct: 503-914-0999 | fax: 954-472-2896 On 11/05/2013 02:09 PM, Coy Cardwell wrote: Hi all. We are using Kamailio 4.0.4 in an Edge Proxy to Central Registrar configuration. The system is in a functional state. Oddly, when we enable TLS, the Kamailio server never responds with a Server Hello to the Client Hello for TLS, so a connection is never established. Has anyone seen anything like this? There are no errors anywhere, the server just never responds to the initial TLS Client Hello. We know the packet arrives and is acknowledged at the TCP level. Just a bit stumped. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamailio dialplan
When you dial 43 you get a prompt or 41? Also, do you see anything in the freeswitch logs or have a sip capture/ Fred Posner | The Palner Group direct: 503-914-0999 | fax: 954-472-2896 On 11/08/2013 06:04 PM, Joli Martinez wrote: I am new to Kamailio and am having an issue with the dialplan setup. I have Kamailio setup as an SBC to handle all user authentication and call routing. I need freeswitch to handle all conferences and voicemails. When I dial 433001 I would like to be transferred to freeswitch for conferences. Right now I have followed the following article and it when I dial 433001 call hangs up and never reaches FS. If I call 43 call does reach FS and I am able to hear FS play the VM prompt. My system is CentoOS 6.4 and FS is installed via yum, but Kamailio is complied. Both FS and Kamailio are on the same box. What commands would you suggest I use to troubleshoot these issues in the future. http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms#dokuwiki__top Also, since I am new could you give some pointers as far as security and documentation. thanks, ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] Kamailio VUC Session on Nov 15
I will be joining the call and IRC but cannot do the hangout. ---Fred > On Nov 14, 2013, at 9:33 AM, Daniel-Constantin Mierla > wrote: > > Hello, > > you, and anyone else that want to join the VUC session, will have to be > tomorrow on irc (channel #vuc on freenode.net) and we will see if we can plug > extra participants in the hangout. It looks very busy, having a large number > (about 10) of developers joining the event and hangout cannot sustain too > many participants. > > However, you can watch the live streaming of the hangout and ask questions > via sip -- this is valid for everyone willing to watch -- VUC video session > will be streamed live, can be watched with a web browser. Questions can be > asked via IRC or audio calls. > > Here are again the links where to look for more information: > > - http://vuc.me > - http://www.kamailio.org/w/2013/10/kamailio-update-on-vuc-nov-15-2013/ > > Cheers, > Daniel > >> On 10/29/13 7:24 PM, Muhammad Shahzad wrote: >> I would like to join this session with google hangout. If that's not >> available then i can call via SIP as well. >> >> Thank you. >> >> >>> On Tue, Oct 29, 2013 at 1:15 PM, Daniel-Constantin Mierla >>> wrote: >>> Hello, >>> >>> we are preparing for a new VUC session to give an update about Kamailio >>> project - a perfect timing as we are just about to release a new major >>> version. >>> >>> We will like to get many developers involved to be able to >>> highlight properly what is new, especially those that did new development >>> for v4.1 - new modules or enhancement to existing modules. However, any >>> developer and community member is welcome to join, we will appreciate it >>> very much, in support of the project. >>> >>> Those that have a google hangout account can participate with video, >>> otherwise there are options to join via sip or pstn audio calls - you can >>> see more details at: >>> >>> - http://vuc.me >>> >>> The number of participants with video is limited, therefore if you plan to >>> do it, let me know to be able to coordinate and send you the invite link >>> when the session starts. >>> >>> No matter you participate with audio/video, you can join the IRC channel >>> #vuc on freenode.net for text chatting during the event. >>> >>> I made quickly a news about the event: >>> >>> - http://www.kamailio.org/w/2013/10/kamailio-update-on-vuc-nov-15-2013/ >>> >>> I will add names of other participants as I get confirmations. Do suggest >>> topics to highlight/discuss there as well. >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.com >>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Trainings - Berlin, Nov 25-28 >>> - more details about Kamailio trainings at http://www.asipto.com - >>> >>> >>> ___ >>> sr-dev mailing list >>> sr-...@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >> >> >> >> -- >> Mit freundlichen Grüßen >> Muhammad Shahzad >> --- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_78...@hotmail.com >> Email: shaherya...@googlemail.com > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Trainings - Berlin, Nov 25-28 > - more details about Kamailio trainings at http://www.asipto.com - > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamdbctl create ?
From the error, it looks like the tables, etc were created and the failure was in granting the privileges. You can always grant them manually, or drop the kamailio database and try the script again. Fred Posner | The Palner Group direct: 503-914-0999 | fax: 954-472-2896 On 11/16/2013 04:10 PM, Thomas Secula wrote: I seem to be in a catch 22, sure it’s something I’ve done wrong. I set dbhost in kamctlrc to be the ip of my remote mysql server I run kamdbctl create and I get root@pcscf:/etc/kamailio# kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database kamailio ... INFO: granting privileges to database kamailio ... ERROR 1044 (42000) at line 1: Access denied for user 'root'@'172.16.101.27' to database 'kamailio' ERROR: granting privileges to database kamailio failed! root@pcscf:/etc/kamailio# Before I run the create from mysql: mysql> select user,host from mysql.user; +--+---+ | user | host | +--+---+ | root | % | | root | 172.16.101.27 | | root | 172.16.101.28 | | root | 172.16.101.29 | | root | localhost | +--+---+ 5 rows in set (0.00 sec) mysql> show grants for 'root'@'%'; +--+ | Grants for root@% | +--+ | GRANT ALL PRIVILEGES ON *.* TO 'root'@'%' IDENTIFIED BY PASSWORD '*FE04940F3F9B339AB9361B2D01AD3D940B215B52' | +--+ 1 row in set (0.00 sec) If I rerun it the create scripts says: root@pcscf:/etc/kamailio# kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database kamailio ... ERROR 1007 (HY000) at line 1: Can't create database 'kamailio'; database exists ERROR: Creating database kamailio failed! root@pcscf:/etc/kamailio# any ideas? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio behind NAT
With a patched version of rtpproxy you can advertise your private ip. http://www.fredposner.com/voip/1457/kamailio-behind-nat/ ---Fred > On Jan 21, 2014, at 6:18 AM, "John Smith" wrote: > > Hello, > > I am currently deploying one Kamailio behind NAT with one Asterisk as > explained in the Asipto KB (Kamailio 4.0.x and Asterisk 11.3.0 using Asterisk > Database). The structure is deployed as described in that document, with the > only addition of one NAT between Kamailio and Internet: > > Phone ———> Nat ———> Kamailio ——> Asterisk > > I have declared the private IP with the advertise option in order to support > the NAT, enabled WITH_NAT and I have installed rtpproxy using standard Debian > package configured as rtpproxy -l public_ip_ -s udp:localhost:7722 > > After setting up two phones which register correctly at Asterisk, I have no > audio at all. > > By placing tcpdumps between nodes I see at Kamailio node both audio from > public IP to internal Kamailio IP and from the latter to the Asterisk IP. In > Asterisk I see audio coming from the Kamailio private IP and then back to the > public IP of the phone. > > My guess is that audio should flow back into Kamailio and then to the phone, > not directly from Asterisk as it is right now. > > Can anyone hint at where I am wrong? > > Thank you > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio behind NAT
On 01/24/2014 09:51 AM, Andrew Pogrebennyk wrote: On 01/23/2014 05:12 PM, Klaus Darilion wrote: It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? If rtpproxy uses only a single listen-IP, then these flags are not needed. Only if you operate rtpproxy in bridge mode, then you need these flags. Bridge mode is necessary if you do not have IP routing between the internal network and the "virtual external" network, or if you want to bridge between IPv4 and IPv6. John, This function can be used to check the direction of every message: http://kamailio.org/docs/modules/4.0.x/modules/rr.html#idp223296 You might also need to append the record-route parameters to remember the flags you have passed to the manage_rtpproxy() initially. Based on the direction of the request and initial flags you can determine what flags to use when calling manage_rtpproxy() for a given in-dialog requests and reply. Hope this helps. Andrew Are the calls being bridged across two interfaces or is the Kamailio just natted? (or is it both?) Fred Posner The Palner Group, Inc. 503-914-0999 (direct) 954-472-2896 (fax) ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Call forwarding from external did
On 02/10/2014 04:00 PM, arun Jayaprakash wrote:
Hello, I have set up call forwarding in Kamailio using user_preference
table. When I make a call from a local extension the callfwd funcition
works. The call gets forwarded to an external did number. The problem
happens when and external call ( from a DID) comes to this extension the
call does not get forwarded. I had to comment out the following lines in
the config file to make it work:
# only local users allowed to call
# if((from_uri!=myself)) {
# sl_send_reply("403", "Not Allowed");
# exit;
# }
My question is if it is a risky think to comment out these lines? If so,
what are my options.
Thank you,
Arun
In short, yes, it is risky-- but in honesty, noone can tell from just
that portion of the config.
You can test this perhaps by trying any number to your system. Is it
also being forwarded?
A good way is to add a flag and check for the presence of that flag if
allowing a non-registered user to make an outside call.
Fred Posner, @qxork
http://palner.com
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[SR-Users] asterisk server as SBC
Hello Daniel,Just got this email from http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb,my requirement is that I am looking at creating an asterisk server to work as SBC.wondering if you can help me on this one. thanks and regards,Fred ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] P-CSCF
Hi all,
we’re just playing around with an IMS setup bases on kamailio. Therefore the
kamailio is used as P-,I- and S-CSCF.
We where able to register two clients through all components.
As we wan to start a call session, the P-CSCF answers with „403 - Forbidden.
You must register with an S-CSCF“…
I found this snippet in kamailio.cfg:
if (!pcscf_is_registered("location")) { send_reply("403","Forbidden - You must
register first with a S-CSCF“); break; }
Can one tell me what exactly the kamailio is checking there? As I figured out
it is looking in database location table.
The table contains the registered users registered towards the IMS.
Another „problem“ we faced with is the rtpproxy. As the ngcp-mediaproxy-ng is
no longer available and replaced with rtpproxy we’re trying to use it. But
kamailio said that the proxy isn’t answering the way expected.
ERROR: rtpengine [rtpengine.c:1622]: rtpp_test(): proxy responded with invalid
response
Any advise would be nice.
Thank you
Fred
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Re: [SR-Users] P-CSCF
Aaron,
thank you. Was a little bit tricky to install it as our slapd isnt in the right
version.. But it seems to work now.. Error messages no longer appear.
Now the message is quiet better:
9(9381) INFO: rtpengine [rtpengine.c:1627]: rtpp_test(): rtp proxy
found, support for it enabled
Again, thank you!
?
Von: sr-users im Auftrag von Aaron
Hamstra
Gesendet: Donnerstag, 28. Januar 2016 21:14
An: Kamailio (SER) - Users Mailing List
Betreff: Re: [SR-Users] P-CSCF
Fred,
I think you would need to use rtpengine instead of rtpproxy.
https://github.com/sipwise/rtpengine
-Aaron
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Fred
Schulz
Sent: Thursday, January 28, 2016 1:24 PM
To: sr-users@lists.sip-router.org
Subject: [SR-Users] P-CSCF
Hi all,
we're just playing around with an IMS setup bases on kamailio. Therefore the
kamailio is used as P-,I- and S-CSCF.
We where able to register two clients through all components.
As we wan to start a call session, the P-CSCF answers with "403 - Forbidden.
You must register with an S-CSCF"...
I found this snippet in kamailio.cfg:
if (!pcscf_is_registered("location"))
{ send_reply("403","Forbidden
- You must register first with a S-CSCF");
break;
}
Can one tell me what exactly the kamailio is checking there? As I figured out
it is looking in database location table.
The table contains the registered users registered towards the IMS.
Another "problem" we faced with is the rtpproxy. As the ngcp-mediaproxy-ng is
no longer available and replaced with rtpproxy we're trying to use it. But
kamailio said that the proxy isn't answering the way expected.
ERROR: rtpengine [rtpengine.c:1622]: rtpp_test(): proxy responded with invalid
response
Any advise would be nice.
Thank you
Fred
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Re: [SR-Users] P-CSCF
Hi Jason,
thank you for your answer. But could you please explain how the UE is
identified?
Is it the contact header? Or some other stuff .. I wasn’t able to find any
information.
And yes, we plan to use the Rx as well. I am can make some traces and logs at
Monday.
Thank you
Fred
Am 28.01.2016 um 20:49 schrieb Jason Penton
mailto:jason.pen...@smilecoms.com>>:
Hi Fred,
I can answer the 1st question only as I am not too clued up with mediaproxy
module and server.
The pcscf_is_registered function is used to confirm that the UE you are sending
the request from is actually registered to the IMS. If this is true, then the
P-CSCF will assert the identity and forward the request to I-CSCF or S-CSCF,
depending on state and request type. Normally the interface between UE and
P-CSCF is via ipsec so it's almost a given that any traffic coming in on the
ipsec pipe can be asserted. In the case without IPSEC however, there are
various methods used to make sure the UE that is sending the request is
actually registered to the IMS. There are a few algorithms that can be
configured to check - ie you can check the contact host and port (this is only
works for invites as MESSAGEs don't have contact headers generally), then you
can check the received IP and port to make sure it's coming from a currently
registered contact that used the same IP:PORT combination, etc, etc.
Alternatively you could remove the check and pass the request onto the S-CSCF
without asserting (not per std though) and then challenge all "cost incurring"
requests with a 407 on the S-CSCF One limitatoin of not getting this
working correctly is that you will not be able to use the Rx interface unless
you can match a contact on the P-CSCF (ims_qos module) but perhaps you are
not interested in Rx interface at the moment?
If you send me a trace (pcap) and logfile I'll take a look as soon as I get a
chance and let you know what the problem is.
Cheers
Jason
On Thu, Jan 28, 2016 at 9:24 PM, Fred Schulz
mailto:fsch...@blackned.de>> wrote:
Hi all,
we’re just playing around with an IMS setup bases on kamailio. Therefore the
kamailio is used as P-,I- and S-CSCF.
We where able to register two clients through all components.
As we wan to start a call session, the P-CSCF answers with „403 - Forbidden.
You must register with an S-CSCF“…
I found this snippet in kamailio.cfg:
if (!pcscf_is_registered("location")) { send_reply("403","Forbidden - You must
register first with a S-CSCF“); break; }
Can one tell me what exactly the kamailio is checking there? As I figured out
it is looking in database location table.
The table contains the registered users registered towards the IMS.
Another „problem“ we faced with is the rtpproxy. As the ngcp-mediaproxy-ng is
no longer available and replaced with rtpproxy we’re trying to use it. But
kamailio said that the proxy isn’t answering the way expected.
ERROR: rtpengine [rtpengine.c:1622]: rtpp_test(): proxy responded with invalid
response
Any advise would be nice.
Thank you
Fred
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--
Jason Penton
Senior Manager: Applications and Services
Smile Communications Pty (Ltd)
Mobile: +27 (0) 83 283 7000
Skype: jason.barry.penton
jason.pen...@smilecoms.com<mailto:name.surn...@smilecoms.com>
www.smilecoms.com<http://www.smilecoms.com/>
[http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg]
This email is subject to the disclaimer of Smile Communications at
http://www.smilecoms.com/home/email-disclaimer/<http://www.smilecoms.com/disclaimer>
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Re: [SR-Users] Facing issue in Registration of SIP Client
Hi, the X-Lite is a normal Sip Client .. for IMS registration you need an IMS client .. such as Boghe based upon doubango. You are only able to use normal username/password register while using X-Lite Cheers, Fred Am 18.02.2016 um 11:53 schrieb sainath.ellend...@wipro.com<mailto:sainath.ellend...@wipro.com>: Hi Franz, Thanks for your response. I have modified the Xlite configuration. Now user is failed to REGISTER with "401 unauthorized - challenging the UE" to challenge the user, here user is not sending any response with authorized details. Please find the attached traces and let me know any more changes has to done. Thanks in advance!! Regards, Sainath From: Franz Edler mailto:franz.ed...@technikum-wien.at>> Sent: 18 February 2016 01:52 To: Sainath Ellendula (NEP); sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org> Subject: RE: Facing issue in Registration of SIP Client Hi Sainath, just check the trace protocol and look at the REGISTER request sent by your Xlite-client. It uses the public identity “sip:bob%40net1.t...@net1.test”. I don’t think that such a user has been provisioned in HSS. The diameter UAA request (packet #436) also clearly says: AVP: DIAMETER_ERROR_USER_UNKNOWN. BR Franz From: sainath.ellend...@wipro.com<mailto:sainath.ellend...@wipro.com> [mailto:sainath.ellend...@wipro.com] Sent: Tuesday, February 16, 2016 1:58 PM To: franz.ed...@technikum-wien.at<mailto:franz.ed...@technikum-wien.at>; sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org> Subject: Reg: Facing issue in Registration of SIP Client Hi Franz, On Successful completion of Registration of IMS client. Now, I was trying to REGISTER SIP Client, But REGISTER request failing with 403 forbidden- User unknown. Please find attachment of X-lite configuration and Tcpdump from #422. Could you please suggest me any changes has to be done. Thanks in advance!! Regards, Sainath The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com<http://www.wipro.com/> The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com<http://www.wipro.com/> ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio 5.0 - B2BUA
On 03/02/2016 12:45 PM, Alex Balashov wrote: > Hi, > > I wanted to raise the possibility of an inline signalling-only B2BUA > component to Kamailio. > +1 for this... for many reasons including seeing what happens when the following threshold is met: > I myself am philosophically opposed to a B2BUA in Kamailio > to the threshold of physical violence. > > -- Alex > Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-224-334-FRED (3733) direct ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] KAMAILIO Installation Problem
On 03/04/2016 01:47 PM, Ed Todd wrote: > I have a local area network with: [snip] > I think the problem is that the reply to the PC is > being sent to 192.168.1.12 instead of 92.13.147.87. If it's on a LAN, I'd assume it would be ok to use the NAT address other than the external address... but that being said, if you're using public IP for everything on the LAN, look into advertised_address: https://www.kamailio.org/wiki/cookbooks/4.3.x/core#advertised_address This being said, I'm confused by your scenario. Fred Posner The Palner Group, Inc. http://www.palner.com (web) ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Git clone failing?
On 03/23/2016 09:45 AM, Richard Good wrote: > Hi > > Is anyone else struggling with git clone? > > richard@richard-laptop-new:~/Smile-dev/git_code$ git clone --depth 1 > --no-single-branch git://git.kamailio.org/kamailio > <http://git.kamailio.org/kamailio> kamailio > Cloning into 'kamailio'... > fatal: read error: Connection reset by peer > Try resetting the peer... I know you're already on the git.kamailio.org site, but you can try: git remote set-url origin https://github.com/kamailio/kamailio.git You can always set it back. I had this problem once, and it resolved once I switched. Switching back hasn't seen the issue return. http://www.fredposner.com/1680/kamailio-4-2-3-update-from-git/ Fred Posner The Palner Group, Inc. +1-224-334-FRED (3733) direct ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] set_advertised_address() / set_advertised_port() + PVs?
On 03/23/2016 01:51 PM, Alex Balashov wrote:
> Hello,
>
> I have a need to deploy Kamailio in AWS in a scenario of this sort:
>
>
>[Public Internet] <---> Kamailio <---> (Internal AWS servers)
>
> In such a scenario, Kamailio would be multihomed.
>
>
> 1. Is this sane? Any unforeseen effects, e.g. vis-a-vis RR, provided
> enable_double_rr is enabled and that two genuinely different network
> interfaces are used?
>
I'm not sure if it's sane, but I do this as well.
> 2. Do set_advertised_address()/set_advertised_port() accept PV
> arguments, or are they pre-PV "core function folk traditions" in the
> same way as rewritehostport() and force_send_socket()?
>
I have a main listen=udp:192.168.25.31 advertise PUBLIC:5060
and then when needed...
set_advertised_address("192.168.25.31");
as in...
if ($rd=~"192.168.25") {
set_advertised_address("192.168.25.31");
}
--fred
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[SR-Users] irc devel meeting summary
Greetings, Minutes from the recent devel meeting have been posted to the wiki: https://www.kamailio.org/wiki/devel/irc-meetings/2016a-minutes In summary, this was a 3+ hour meeting held on IRC #kamailio -- one of the longer meetings on record. No current "critical" issues were open, and the meeting moved to a discussion of for a testing framework and being able to load as many modules as possible in a running Kamailio system. Olle Johansson suggested a competition for Kamailio World: "The one with a kamailio with most loaded modules running at kamailio-world gets a beer." A discussion then occurred regarding the appropriateness of returning a 200 OK when no data exists during a xmlrpc request. The consensus seemed to side with how we currently return data with the suggestion of moving further discussion to the mailing list. The Kamailio team will be looking to both upgrading the Kamailio servers to Debian Jessie as well as utilizing a "responsive" template for the main website; making the site more mobile/user friendly. Kamailio 5 was discussed with a suggestion for a developers meeting in Stockholm, perhaps in June, to hammer out the framework. Daniel was thanked for moving forward with the Lua routing aspects of Kamailio 5 while ensuring that you will still be able to utilize the config file (as current) without Lua. Additional languages, such as python, will be coming. Daniel will be working on a tutorial for exporting the kamailio.cfg to an embedded interpreter in the future. As part of Kamailio 5, the source tree structure will be improved, most likely utilizing subdirectories to better organize the different elements of the software. There will also be a change in the method of checking the database schema for compatibility with the version of software running. This summary represents just an appetizer of the incredible meeting that took place. The minutes are available for you to read at your leisure and your participation is always encouraged through the mailing lists, IRC, as well as github. The meeting never ends... it just continues on the mailing lists. =) Best regards, Fred Posner http://www.palner.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] LCR Module Question
Have you considered either dispatcher or just using a failure route? -- Fred > On May 6, 2016, at 7:15 AM, Alberto Sagredo > wrote: > > Hi > > I have it working but i have re-read documentation and do not see how to do > what i need. > > I explain it :) > > Now i have only one LCR provider and i need to add a backup one. > > I do not know if its enough to add under same lcr_id or its better to add > with different one and add several lcr_rule and lcr_rule_target > > Acordding to next_gw() function, what is better to be used? In my case. a > primary providers is used and only a backup one for the momment. > > Thanks for your help > > Alberto > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SRV and dialog stickiness
On 05/25/2016 04:32 PM, Alex Balashov wrote: > Hello, > > Sorry if this is a tired, worn question, but I've not dealt much with > Kamailio's SRV support before: > > If a registrant has a contact binding whose domain component is subject > to an SRV lookup with load-balanced or weighted entries, how does one > solve the problem of ensuring that subsequent in-dialog requests go to > the same host as the initial INVITE? Does Kamailio offer some facility > for doing this? Is it somehow accommodated by SIP? > > -- Alex > Clarification: Why would this be wanted? I ask, as if you were using a srv record as the result of a load balance lookup, wouldn't the point be to be able to quickly change location of the domain in case of an outage/issue? Otherwise, I'm not positive the benefit of doing a srv lookup for this scenario. -- Fred Posner @fredposner The Palner Group, Inc. http://www.palner.com direct/sms: +1 (503) 914-0999 direct/sms: +1 (224) 334-FRED ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] username of Contact header
On 05/27/2016 10:21 AM, Al S wrote:
> Hi,
>
> I am trying to read username portion of Contact header:
>
> My Contact header content has a URI value such as:
> sip:813111@10.10.10.10.:5060
>
> and I am trying to read : 813111
>
> I tried the following perl similar RE and it didn't work:
>
> ($var(main_number2)) = $ct =~ /sip:(.*)@/;
>
> Thanks,
> AS
Check out transformations:
$var(user) = $(ct{tobody.user});
http://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#uri_transformations
http://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#to-body_transformations
--
Fred Posner
@fredposner
The Palner Group, Inc.
http://www.palner.com
direct/sms: +1 (503) 914-0999
direct/sms: +1 (224) 334-FRED
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Re: [SR-Users] High availability
If it's just 2 servers, consider as Juha said, corosync/pacemaker with drbd. Fred Posner direct: +1 (224) 334-FRED (3733) > On Jun 5, 2016, at 5:26 PM, Moacir Ferreira > wrote: > > Hi, > > Sorry... I should have mentioned before. You guys are thinking on the > standard Internet SIP calls' behavior while I am trying to use Kamailio on a > large "industrial" project. This said: > > Assuming that the end-point is "smart", the DNS method is functional but it > would take quite a while before the UA (phone) recovers from the previous > name/IP binding it has in cache; > SRV is good for a "smart" UA that, unfortunately, is not the case; > Same for the phone units as they are industrial "Help Points" and so quite > "dummy". > > While I never tested it, I thought I could use two Kamailio servers with a > mysql cluster like mariadb-galera where, for Kamailio functions, one server > would be "active" and another "passive" server. Then use keepalived for > monitoring the "active" Kamailio and starting the "passive" server if the > active Kamailio fails. Without any testing, tests that I think I should have > done before putting questions in here, my questions are: > > Suppose that I have two Kamailio servers, one "active" and another one > "passive" (not running) where the mysql databases are synchronized in between > two servers using MySQL Galera. Using keepalived I would monitor the active > Kamailio instance. Should it fails, start the "passive" Kamailio instance > using the same MySQL database that were supposed to be synchronized. Would > this new Kamailio instance be able to find a called number? Why this > question? As long as I understand, Kamailio will always challenge the UA for > authentication before making a call, so if this second server gets a call > request it would just challenge and authenticate the caller. The "key point" > would be having this new Kamailio instance aware about the called > destinations. So, delivering a MySQL database, with the latest data the > active Kamailio had, to this new Kamailio instance would be enough to allow > it find the called party? > > Anyway, can you guys comment on my "thoughts"? Is it possible? Am I missing > something? Would you suggest another approach for such scenario? > > Cheers! > Moacir > > Date: Sun, 5 Jun 2016 21:07:41 +0200 > From: chabert.loic...@gmail.com > To: sr-users@lists.sip-router.org > Subject: Re: [SR-Users] High availability > > Hello Bill, > > I have made kamailio ha using exabgp with loopbacks. > > Check https://github.com/Exa-Networks/exabgp > > With bgp, kamailio cluster can be splited on severals datacenters. > > Regards. > > Le 5 juin 2016 20:53, "Bill" a écrit : > Hi Moacir > > We have only found three ways to handle failover. > 1. Change the DNS entry whenever a failure is detected. > 2. Use SRV records to display an alternate route. > 3. Use the failover mechanism in the phone itself > > 1. works, but it may take some time for your ua's to become aware of the > change > 2. never have been able to get this to work as advertised. > 3. Works pretty well depending on the phone. (We use mostly Yealink's and > they seem to handle the failover pretty well.) > > Hope this helps > > On 06/05/2016 07:41 AM, Moacir Ferreira wrote: > Hi, > > I got two questions regarding high availability: > > 1 - Should my Kamailio server fail, I would like another Kamailio > "box/server" to take over with minimum services disruption. What is the > "community" advice for such environment? > > 2 - Should my main PSTN gateway fail, what would be the best mechanism to > redirect calls to a second PSTN gateway? > > Cheers! > Moacir > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > ___ SIP Express Router (SER) and > Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Why SDPOPS does not remove attributes in SDP
On 06/17/2016 09:25 AM, Richard Fuchs wrote:
> On 17/06/16 03:46 AM, Dmitry wrote:
>> Hi all
>> I have the following code:
>>
>> if($T_reply_code=="200")
>> {
>> if(has_body("application/sdp"))
>> {
>> xlog("L_INFO", "RTPENGINE received internal reply
>> $T_reply_code $rr SDP extra lines will be removed");
>>
>> set_rtpengine_set("0");
>> rtpengine_manage();
>> sdp_remove_line_by_prefix("a=rtcp");
>> sdp_remove_line_by_prefix("a=ssrc");
>> sdp_remove_line_by_prefix("a=ice");
>> sdp_remove_line_by_prefix("a=candidate");
>>
>> xlog("L_INFO", "RTPENGINE received internal reply
>> $T_reply_code $rr SDP extra lines removed with SDPOPS");
>>
>> }
>>
>> }
>> When I look through traces - I see that 200 ok(with SDP) has all these
>> attributes and they are not removed.
>>
>> Why SDPOPS does not remove these attributes?
>
> Probably because there's a problem rewriting parts of the SDP body more
> than once. But if you don't want ICE attributes in the output SDP, you
> can use the rtpengine flags ICE=remove. You can influence rtcp-mux
> attributes in the same way. See docs.
>
> Cheers
>
Are your log messages triggered?
--fred
0x6235BD69.asc
Description: application/pgp-keys
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Re: [SR-Users] Video conferencing with Kamailio
On 07/08/2016 12:36 AM, Jay Li wrote: > Dear All, > > I'm curious if anybody has set up an infrastructure for video > conferencing utilizing Kamailio as a proxy (like NAT support and so so). > I found a kind of old tutorial "Run you own Skype-like service in less > than one hour" kamailio:skype-like-service-in-less-than-one-hour [Asipto > - SIP and VoIP Knowledge Base Site] > <http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour> > written > in 2013 on Kamailio 3.1. I wonder if this feature has been further > developed in later releases. I know Kamailio is more often used as a SIP > proxy, but somehow I got the impression that Kamailio supports WebRTC, > so I wonder if it's possible to implement a many to many video > conferencing infrastructure using Kamailio. If the answer is yes, how's > the performance or the bottleneck I should pay more attention to (maybe > bandwidth for video traffic, especially with NAT? ) Anyway I'm a newbie > to both Kamailio and video conference, so any suggestion/discussion is > appreciated. Thanks. > > Regards, > Jay > Jay, The principles in the tutorial hold true today, and yes, kamailio does support WebRTC. Assuming you want your video conference to have more that two parties, you will also need a media server in addition to Kamailio. Kamailio does not handle the mixing, timing, etc of media to enable multi-person video conferencing. So, you will need this being done either by a separate media server or endpoint capable of doing this. There are some products like Jitsi Video Bridge and FreeSWITCH that support video conferencing "out of the box." You can combine these with Kamailio as well to handle additional security, authentication, etc. -- Fred Posner @fredposner http://palner.com direct/sms: +1 (503) 914-0999 direct/sms: +1 (224) 334-FRED 0x42AE1A40.asc Description: application/pgp-keys ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Video conferencing with Kamailio
On 07/11/2016 11:40 AM, Jay Li wrote: > Fred, > > Thanks a lot your detailed explanation. About the media server addition > to Kamailio, do you have any suggestions I should look into besides > Jitsi and FreeSWITCH? Thanks. > > Regards, > Jay You could look into Asterisk as well, but I've not used it for video conferencing so cannot speak from experience. -- Fred Posner @fredposner http://palner.com direct/sms: +1 (503) 914-0999 direct/sms: +1 (224) 334-FRED 0x42AE1A40.asc Description: application/pgp-keys ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Commercial SBC or Kamailio
On 09/15/2016 10:38 AM, Linux Vince wrote: > We are trying to setup VOIP infrastructure, mainly wholesale and retail. > > What is best option if money is not the problem? > > Developing our own infrastructure using Kamailio and other open source > packages to act as SBC and switch or buy a commercial solution like > GenBand/Sonus/Sansay. > > We are looking for high performance with scalability to handle thousands > of call setups per second over the period of few years. > > I have no idea on how/if can kamailio outperform commercial solutions or > not. > > Advantage of using kamailio is flexibility and possibility of > customization as per requirement, > > I am new to this list so please advise if this is not a valid question > to be asked here. > Normally, we try to keep commercial discussions on the business list: http://lists.kamailio.org/cgi-bin/mailman/listinfo/business You also may want to check out the business directory: https://www.kamailio.org/w/business-directory/ There are some products that involve Kamailio at it's core, such as Canonical SIP Routing Platform (CSRP), Enswitch, Sip:Wise, 2600hz, etc. -- Fred Posner @fredposner The Palner Group, Inc. http://www.palner.com direct/sms: +1 (503) 914-0999 direct/sms: +1 (224) 334-FRED ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Size of passwords fields in db
On 10/05/2016 10:25 AM, Daniel-Constantin Mierla wrote: > Hello, > > writing here to decide on a topic opened by pull request 779: > > - https://github.com/kamailio/kamailio/pull/779 > > what would be a fair size for db column storing a password that one > would like to have for proper security? > > I would like to make it consistent over all tables that have a password > column by defining a xml entity for the size of these columns. The pull > request suggests 64 chars, has anyone other opinions on making it larger > or smaller? > > If they are defined varchar, then should not be a problem of allocated > size, so we can go with 128 if someone feels it worth doing larger now > so we don't have to change it again in the near future. > > This change is about db schema, the modules I expect to work with > allocated strings (or have length checks) in this case and should not be > affected. > > Cheers, > Daniel > Although I can see why someone might consider the need for larger than varchar 64, I really don't see a need for it. Assuming if you needed more characters it would probably be time to use additional authentication methods. I believe Polycom still max's out at 32. --fred 0x42AE1A40.asc Description: application/pgp-keys ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Size of passwords fields in db
On 10/06/2016 05:43 AM, Daniel-Constantin Mierla wrote: > On 05/10/16 16:35, Fred Posner wrote: >> On 10/05/2016 10:25 AM, Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> writing here to decide on a topic opened by pull request 779: >>> >>> - https://github.com/kamailio/kamailio/pull/779 >>> >>> what would be a fair size for db column storing a password that one >>> would like to have for proper security? >>> >>> I would like to make it consistent over all tables that have a password >>> column by defining a xml entity for the size of these columns. The pull >>> request suggests 64 chars, has anyone other opinions on making it larger >>> or smaller? >>> >>> If they are defined varchar, then should not be a problem of allocated >>> size, so we can go with 128 if someone feels it worth doing larger now >>> so we don't have to change it again in the near future. >>> >>> This change is about db schema, the modules I expect to work with >>> allocated strings (or have length checks) in this case and should not be >>> affected. >>> >>> Cheers, >>> Daniel >>> >> Although I can see why someone might consider the need for larger than >> varchar 64, I really don't see a need for it. Assuming if you needed >> more characters it would probably be time to use additional >> authentication methods. > That refreshed my mind that we have now support for sha 256, which means > that ha1 fields need to be 64 (and they are now), but wondering if > someone will want to have and add sha 512 any time soon, which means the > ha1 fields need to be 128... I guess there's really no cost to increasing it to 128 / sha 256 and give ourselves some good time before we need to reconsider. The storage cost will still be based on the actual value used and not the size constraint; giving people options. >> >> I believe Polycom still max's out at 32. > I haven't looked at phones restrictions on this, but I saw old devices > accepting only digit based passwords... hopefully not many out there at > this moment. > > A stronger auth method would be highly desired, but I guess a lot of > people are restricted by deployed devices. > > Cheers, > Daniel > --fred ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Asterisk Security Advisory (AST-2016-009)
Thank you for the post-- definitely appreciate you sharing it on this list. --fred On 12/8/16 6:02 PM, Matthew Jordan wrote: Hey all - The Asterisk project just released a security advisory for a security vulnerability in which Asterisk using chan_sip with a proxy can allow for unauthenticated calls. This affects all supported versions of Asterisk (11, 13, 14). Since that may be relevant to those on this mailing list who are not also on the asterisk-users mailing list, I thought it prudent to mention it here as well. A description of the vulnerability follows: Description The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue does not affect you. The announcement can be seen here: http://lists.digium.com/pipermail/asterisk-announce/2016-December/000662.html Thanks again to Walter Doekes for reporting the vulnerability and providing the patch to fix it. Matt ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.
> listen=udp:MY_IP_ADDR:5060 advertise MY_PUBLICIP_ADDR:5060 That statement does not exist anywhere in the files you sent. --fred On 12/29/2016 11:19 AM, Pranathi Venkatayogi wrote: > Yes. I defined advertised address and even used listen with advertise as > below. Still Kamailio does not send publicip in record route header. > listen=udp:MY_IP_ADDR:5060 advertise MY_PUBLICIP_ADDR:5060 > > > -Original Message- > From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of > Daniel Grotti > Sent: Thursday, December 29, 2016 6:31 AM > To: sr-users@lists.sip-router.org > Subject: Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised > public IP. > > Hi, > not sure if I understood it right but, have you defined the > advertised_address ? That should be used in Via and RR as well: > > https://www.kamailio.org/wiki/cookbooks/4.4.x/core#advertised_address > > > Daniel > > > On 12/29/2016 12:09 AM, Pranathi Venkatayogi wrote: >> I implemented full NAT logic as per the sample config. Still unable to >> resolve the issue. >> >> How do I let Kamailio change record_route header to use public ip address? >> >> >> >> Please help!!! >> >> >> >> (attached are latest scripts) >> >> >> >> *From:* Pranathi Venkatayogi >> *Sent:* Wednesday, December 28, 2016 12:39 PM >> *To:* 'sr-users@lists.sip-router.org' >> *Subject:* Kamailio behind NAT, ACK to private IP not advertised public IP. >> >> >> >> Hi, >> >> I am encountering the same problem described in google groups >> <https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. >> However I dint not find any resolution hence writing again. >> >> >> >> 200 OK sent from the server has private Ip in its record route. As >> you see below, though the message is received on public IP >> (63.149.103.72) , the record route is set to private IP >> (172.31.211.31) >> >> I used listen with advertise of public IP, it did not work. Please >> find attached the config I am using. >> >> >> >> How do I change it send public ip only when talking to external world. >> >> Can someone point to me clear documentation how to configure >> Kamailio for NAT traversal. >> >> >> >> *The following message is sent from Kamailio behind NAT to the public >> computer.* >> >> 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, >> +0:08:42.690309 >> >> 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 >> >> SIP/2.0 200 OK >> >> Via: SIP/2.0/TLS >> 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e38 >> 1a96979945bd969989ffe9dca3a9;alias >> >> Record-Route: > > >> >> Call-ID: eb8670eec4354acdb69fd26f5625b75c >> >> From: "cust1" >> ;tag=2f25d2ae690747c48c874 >> e0b415ca03c >> >> To: >> ;tag=1c33ad41f6f44cae8ae >> 8e060f30fe119 >> >> CSeq: 4665 INVITE >> >> Server: Blink 3.0.0 (Windows) >> >> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, >> MESSAGE, REFER >> >> Contact: >> >> Supported: 100rel, replaces, norefersub, gruu >> >> Content-Type: application/sdp >> >> Content-Length: 355 >> >> v=0 >> >> o=- 3691844303 3691844304 IN IP4 10.0.27.108 >> >> s=Blink 3.0.0 (Windows) >> >> t=0 0 >> >> m=message 2855 TCP/TLS/MSRP * >> >> c=IN IP4 10.0.27.108 >> >> a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp >> >> a=accept-types:message/cpim text/* image/* >> application/im-iscomposing+xml >> >> a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml >> >> a=setup:active >> >> -- >> >> >> >> *The following is the ACK sent by public computer in reply to the >> above message. Note this message never reaches the Kamailio server as >> it is sent to private IP.* >> >> 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, >> +0:08:42.690309 >> >> 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 >> >> ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 >> >> Via: SIP/2.0/TLS >> 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2; >> alias >> >> Max-Forwards: 70 >> >> From: "cust1" >> ;tag=2f25d2ae690747c48c874 >> e0b415ca03c >&
Re: [SR-Users] Kamailio not processing SIP TCP
What happens when you try:
modparam("sipcapture", "hep_capture_on", 1)
On 01/13/2017 10:33 AM, JR Richardson wrote:
> Iptables is not blocking, but it was worth a check.
>
> Thanks.
>
> JR
>
>
> I assume you have ruled out firewall? It's something that can nab even
> experienced people:
>
> # iptables -Ln
>
> -- Alex
>
> On Thu, Jan 12, 2017 at 03:25:27PM -0600, JR Richardson wrote:
>
>> Hi All,
>>
>> Just enabled SIP TCP on a homer capture server, I can see the SIP TCP
>> Sessions on the server with ngrep, just like all the UDP traffic. I
>> have Kamailio listening on TCP ports but its not capturing any TCP
>> traffic.
>>
>> kamailio.cfg:
>>
>> #disable_tcp=yes
>> listen=tcp:10.99.99.99:5060#monitor port
>> listen=udp:10.99.99.99:5060 #monitor port
>>
>> loadmodule "pv.so"
>> loadmodule "db_mysql.so"
>> loadmodule "sipcapture.so"
>> loadmodule "textops.so"
>> loadmodule "rtimer.so"
>> loadmodule "xlog.so"
>> loadmodule "sqlops.so"
>> loadmodule "htable.so"
>> loadmodule "sl.so"
>> loadmodule "siputils.so"
>>
>>
>> modparam("sipcapture", "capture_on", 1)
>> modparam("sipcapture", "hep_capture_on", 0)
>> modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070")
>> modparam("sipcapture", "raw_interface", "eth1")
>> modparam("sipcapture", "raw_ipip_capture_on", 0)
>> modparam("sipcapture", "table_name", "sip_capture")
>> modparam("sipcapture", "raw_sock_children", 4)
>> modparam("sipcapture", "db_insert_mode", 0)
>> modparam("sipcapture", "raw_moni_capture_on", 1)
>> modparam("sipcapture", "promiscious_on", 1)
>> modparam("sipcapture", "raw_moni_bpf_on", 1)
>> modparam("sipcapture", "capture_node", "homer02")
>> modparam("sipcapture", "authorization_column", "authorization")
>>
>>
>> ## logging all INVITES top of the [route] block
>> if (is_method("INVITE|REGISTER")) {
>> xlog("L_INFO", "Received INVITE \"$fU\" to \"$rU\"
>> from \"$si\"\n");
>>
>> Logging reports all SIP UDP traffic to logs fine, but no TCP traffic.
>>
>> root@homer02:~# netstat -al
>> Active Internet connections (servers and established)
>> Proto Recv-Q Send-Q Local Address Foreign Address State
>> tcp0 0 homer02.me.com:sip*:* LISTEN
>>
>>
>> I don't think this is a homer issue because logging invites is prior
>> to any homer processing. I'm thinking this is something simple I'm
>> overlooking, any help is much appreciated.
>>
>> Thanks.
>>
>> JR
>> --
>> JR Richardson
>> Engineering for the Masses
>> Chasing the Azeotrope
>>
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Re: [SR-Users] Phone number formats per country
Normally I run this against a carrier rate sheet, using the description. Both Twilio and Flowroute have decent download-able sheets with prefix <-> country/mobile description. --fred On 01/18/2017 09:21 AM, Daniel-Constantin Mierla wrote: > Hello, > > slightly off-topic, but related to voip -- does anyone have a link that > can quickly share to a site with up to date details about phone > numbering plans for as many country as possible? > > Ideally to include the toll and premium number prefixes, the split > between mobile and fixed lines if it is the case. > > Cheers, > Daniel > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Replacing an ACME Packet Net-Net SBC
Alex's article is one of my favorites. That being said, we switched out an Acme SBC for openser (at the time) and was immediately thrilled. Fred Posner The Palner Group, Inc. 503-914-0999 (direct) 954-472-2896 (fax) On 02/20/2014 01:14 PM, Alex Balashov wrote: Francesco, Have a look at this blog post: http://www.likewise.am/2013/03/kamailio-as-an-sbc-session-border-controller/ That said, I agree with Carsten's suggestion of SEMS. On 02/20/2014 11:04 AM, Francesco Maria Magnini wrote: Hi, I would like to have some suggestions about a full replacement of an ACME Packet Net-Net Session Border Controller. By now, ACME SBC performs all the SBC functionalities, mainly: - it is used as a SIP endpoint for SIP client registrations - it is used as a SIP endpoint for interconnection to multiple SIP carriers via SIP trunks - it is used for NAT traversal In this deployment, the SIP Server communicates only with the SBC and this one takes care of the communication between the SIP Server and the external SIP entities (UA clients, SIP Trunks). In this scenario, can I consider to replace the SBC with Kamailio? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Replacing an ACME Packet Net-Net SBC
On 2/20/14, 5:55 PM, Francesco Maria Magnini wrote: @Carsten I looked at http://www.iptel.org/sems and seems to be only broken links to downloads. Do you know if the project is still maintained? @Fred Are you using openser as a B2BUA? No, because of course Kamailio is not a b2bua. =) In the case of the ACME replacement, we used it to: - handle NAT (rtpproxy) - user regs - load balance - lcr - security - routing and some other little hacks. -- Fred Posner | The Palner Group, Inc. http://qxork.com Il giorno 20/feb/2014, alle ore 19:42, Fred Posner ha scritto: Alex's article is one of my favorites. That being said, we switched out an Acme SBC for openser (at the time) and was immediately thrilled. Fred Posner The Palner Group, Inc. 503-914-0999 (direct) 954-472-2896 (fax) On 02/20/2014 01:14 PM, Alex Balashov wrote: Francesco, Have a look at this blog post: http://www.likewise.am/2013/03/kamailio-as-an-sbc-session-border-controller/ That said, I agree with Carsten's suggestion of SEMS. On 02/20/2014 11:04 AM, Francesco Maria Magnini wrote: Hi, I would like to have some suggestions about a full replacement of an ACME Packet Net-Net Session Border Controller. By now, ACME SBC performs all the SBC functionalities, mainly: - it is used as a SIP endpoint for SIP client registrations - it is used as a SIP endpoint for interconnection to multiple SIP carriers via SIP trunks - it is used for NAT traversal In this deployment, the SIP Server communicates only with the SBC and this one takes care of the communication between the SIP Server and the external SIP entities (UA clients, SIP Trunks). In this scenario, can I consider to replace the SBC with Kamailio? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Replacing an ACME Packet Net-Net SBC
On 2/20/14, 6:25 PM, Francesco Maria Magnini wrote: Fred, in you ACME replacement, kamailio doesn’t rewrite headers for handling RTP/SIGNALING and stay in the middle? For nat it did. For others the media server did. You can easily force all connections to use rtpproxy to do what you ask. We chose to do this for NAT and for media servers outside of kamailio to be the other choice. All calls were on one, the other, or both. -- Fred Posner | The Palner Group, Inc. http://qxork.com Il giorno 21/feb/2014, alle ore 00:03, Fred Posner mailto:f...@palner.com>> ha scritto: On 2/20/14, 5:55 PM, Francesco Maria Magnini wrote: @Carsten I looked athttp://www.iptel.org/semsand seems to be only broken links to downloads. Do you know if the project is still maintained? @Fred Are you using openser as a B2BUA? No, because of course Kamailio is not a b2bua. =) In the case of the ACME replacement, we used it to: - handle NAT (rtpproxy) - user regs - load balance - lcr - security - routing and some other little hacks. -- Fred Posner | The Palner Group, Inc. http://qxork.com <http://qxork.com/> Il giorno 20/feb/2014, alle ore 19:42, Fred Posner mailto:f...@palner.com>> ha scritto: Alex's article is one of my favorites. That being said, we switched out an Acme SBC for openser (at the time) and was immediately thrilled. Fred Posner The Palner Group, Inc. 503-914-0999 (direct) 954-472-2896 (fax) On 02/20/2014 01:14 PM, Alex Balashov wrote: Francesco, Have a look at this blog post: http://www.likewise.am/2013/03/kamailio-as-an-sbc-session-border-controller/ That said, I agree with Carsten's suggestion of SEMS. On 02/20/2014 11:04 AM, Francesco Maria Magnini wrote: Hi, I would like to have some suggestions about a full replacement of an ACME Packet Net-Net Session Border Controller. By now, ACME SBC performs all the SBC functionalities, mainly: - it is used as a SIP endpoint for SIP client registrations - it is used as a SIP endpoint for interconnection to multiple SIP carriers via SIP trunks - it is used for NAT traversal In this deployment, the SIP Server communicates only with the SBC and this one takes care of the communication between the SIP Server and the external SIP entities (UA clients, SIP Trunks). In this scenario, can I consider to replace the SBC with Kamailio? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Configuring Kamailio as an upstream proxy for FreeSwitch and which RTP proxy to choose
On Mon Feb 24 10:50:07 CET 2014 Sean Kemball wrote: > New to Kamailio and FreeSwitch, loosely familiar with SIP mechanics, > and not a complete network idiot... but please be gentle. :) Welcome!s, > Questions: > > 1.Should the proposed topology, with Kamailio + an RTP proxy > behind a firewall, relaying to FS on an inside interface, work? > (Can't see why not) Yes, you said that your upstream is on the same private network. So it should be pretty straight forward. > 2.Does it need a local RTP proxy on the Kamailio box, particularly > if we turn off the ASA SIP inspect stuff? If you are all on the same private network, I would let FreeSWITCH handle the RTP, but you can do this a variety of ways. > 3.Can you recommend which RTP proxy to use? There seem to be at > least 3 that work with Kamailio. The box is CentOS 6.5, and it would > be nice to use known-to-work packages rather than compile from source. > (But eh, if I haveta). On your scenario, I'd just use FreeSWITCH for the media proxy. Again, many different ways to go here. > 4.Can anyone point me to some docs to explain what ports need to > be open between the Kamailio box and my upstream proxy/media server? > I can be more liberal between inside and DMZ I guess. Your upstream provider would generally tell you which rtp ports they would want opened. > 5.Is static NAT in this environment going to bite me, or should > it be OK? I've never had an upstream provider communicate with me on private nat. > 6.Is there any better documentation that we should be using to > make this easier, or should I just man up and try harder? Man up. =) Practive makes perfect. -- Fred Posner The Palner Group, Inc. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] fresh kamailio installation unable to connect through jitsi
On 02/26/2014 08:52 PM, Michelle Jun wrote: hi i just finished installation of kamailio on centos 6 64 bit following the docs on http://www.fredposner.com/voip/1457/kamailio-behind-nat/ the user able to chat through jitsi just fine, but when calling, one user able to connect, while the other showing only "connecting" (audio/video) any idea what did i do wrong? here is the /var/log./messages thank you If you're natted, make sure you have your firewall forwarded for the ports you've selected for rtp and sip. Fred Posner The Palner Group, Inc. 503-914-0999 (direct) 954-472-2896 (fax) ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] fresh kamailio installation unable to connect through jitsi
On 02/26/2014 09:11 PM, Michelle Jun wrote: hi Fred yes, i forwarded both TCP/UDP 5060 dan 2-3 like in your blog but still having the issue thanks The rtp forwarding should be just udp. For the sip, that's up to how you're making the connections. Did you specify a range when you started the rtpproxy? Do you have any of the sip traffic (from ngrep, etc.)? --fred ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] fresh kamailio installation unable to connect through jitsi
On 2/27/14, 1:21 PM, Michelle Jun wrote: m=audio 21064 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 101. m=video 23134 RTP/AVP 105 99. It does look like it's within the range. I would generally ensure that your firewall is forwarding the ports. -- Fred Posner | The Palner Group, Inc. http://qxork.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Caller ID name
Just to add, besides the uac having some of the best example names...
the callerid you mentioned is most likely set on your phone config;
which kamailio is just passing along.
Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)
On 03/19/2014 07:18 PM, Alex Balashov wrote:
Hello Abdul,
This depends on how you want to signal the caller ID.
If you want to indicate it with the P-Asserted-Identity header, which
overrides both Remote-Party-ID and the 'From' value, you can just append
your own header with a display name value:
append_hf("P-Asserted-Identity: \"SHERIF MALIK\"
\r\n");
If you want to actually override the From display value, that's a bit
more complex, since proxies aren't technically supposed to do that.
However, the 'uac' module gives you this capability:
http://kamailio.org/docs/modules/4.1.x/modules/uac.html#uac.f.uac_replace_from
e.g.
uac_replace_from("\"SHERIF MALIK\"", ""); # Don't modify From URI.
-- Alex
On 03/19/2014 07:15 PM, malik sherif wrote:
Hello,
I set SIP users using kamctl add command with username ( I put the
phone numbers) doamin (doamin name) and password. Is their a way to add
caller id name? when I make a call , i see the phone number but for
caller id name I displayline1.
Thank you for your help.
Abdul
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Re: [SR-Users] NAT Traversal issue
It looks like you may be running Kamailio behind NAT as well, no? Can you provide any traffic on the connections that fail? Fred Posner The Palner Group, Inc. 503-914-0999 (direct) 954-472-2896 (fax) On 04/03/2014 08:44 AM, Ravi wrote: Dear Kamailio'ns, I am awaiting somebody's suggestions/hints/comments on this issue, with that i can proceed further. Please anybody help me in resolving this issue. Any help will mean a lot and greatly appreciate. Regards, Ravi ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] sql_xquery() and xavp checks
When I use xquery, I generally use it with a stored procedure that
returns a value if not matched... so I always return at least one row
with a variable of 'fail', -1, etc. to evaluate.
I like this for a variety of reasons (ie changing sql without changing
the config)... but that being said...
Wouldn't this still work for you:
if($dbr(gateways=>rows)>0) { }
Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)
On 04/05/2014 11:32 AM, Alex Balashov wrote:
Hi,
When using sql_xquery() like this:
sql_xquery("ca", "SELECT * FROM gateways", "gateways");
... what's a good way to check if any rows were returned? Since one does
not have a $dbr(gateways=>rows) value in this scenario, what should one do?
- is_avp_set("$xavp(gateways=>id)")) does not appear to operate on
XAVPs, or at least, the fixup functions reject them:
ERROR: avpops [avpops.c:935]: fixup_is_avp_set(): bad attribute name
<$xavp(gateways=>id)>
- the 'defined' operator does not appear to return a negative condition
here:
if(!defined $xavp(gateways=>id))
This condition evaluates to true.
Much appreciated!
-- Alex
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Re: [SR-Users] sql_xquery() and xavp checks
> I don't think so. As I understood the documentation, at least, $dbr
> doesn't get populated in this case; the rows just go straight to an
> xavp list. I suppose I should verify that.
>
Looks like you're right.
Tested various methods.
Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)
On 04/05/2014 11:45 AM, Alex Balashov wrote:
On 04/05/2014 11:42 AM, Fred Posner wrote:
When I use xquery, I generally use it with a stored procedure that
returns a value if not matched... so I always return at least one row
with a variable of 'fail', -1, etc. to evaluate.
I actually do that too in many cases, but only because I often need to
pass back additional data about what went wrong (for logging) if no row
was found, so returning a row regardless, with a status column (with a
value like -1) and hijacking another column for some kind of
human-readable explanation.
I like this for a variety of reasons (ie changing sql without changing
the config)... but that being said...
Wouldn't this still work for you:
if($dbr(gateways=>rows)>0) { }
I don't think so. As I understood the documentation, at least, $dbr
doesn't get populated in this case; the rows just go straight to an xavp
list. I suppose I should verify that.
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Re: [SR-Users] sql_xquery() and xavp checks
On 04/05/2014 09:01 PM, Alex Balashov wrote: Does that work for SELECT queries? The documentation says it's only for INSERT, UPDATE and DELETE. It did not during my test this afternoon. --fred On 6 April 2014 02:14:24 CEST, Kelvin Chua wrote: dunno if this helps but i use $sqlrows(ca) to check whether there are rows returned Kelvin Chua On Sat, Apr 5, 2014 at 9:15 AM, Alex Balashov mailto:abalas...@evaristesys.com>> wrote: On 04/05/2014 12:14 PM, Fred Posner wrote: > I don't think so. As I understood the documentation, at least, $dbr > doesn't get populated in this case; the rows just go straight to an > xavp list. I suppose I should verify that. > Looks like you're right. Tested various methods. This does make me wonder if there is a "leak" of result handles here, since sql_result_free() is not possible, but presumably there's an internal result handle still getting allocated. Or maybe that was for the memory allocated to the $dbr(...) data only and independent of the underlying DB API. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sent from my mobile, and thus lacking in the refinement one might expect from a fully fledged keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] sql_xquery() and xavp checks
Have you tried something like...
if (sql_xquery("ca", "SELECT * FROM gateways", "gateways") == 1) {
#do stuff
} else {
#dang nabbit
}
Fred Posner
The Palner Group, Inc.
503-914-0999 (direct)
954-472-2896 (fax)
On 04/06/2014 02:37 PM, Alex Balashov wrote:
On 04/06/2014 12:37 AM, Kelvin Chua wrote:
$dbr for SELECTs
Unless it's not, because you're using sql_xquery(). :-)
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Re: [SR-Users] Adding dialogs to a profile in a failure_route
On 04/28/2014 01:49 PM, Alex Balashov wrote: Hi, [SNIP] So, the question is, am I doing something wrong? What's the best way to accommodate this scenario? I don't know if I want to track the dialog or add it to a profile until after I get the 302. Out of curiosity, if you call set_dlg_profile() at the initial invite and then so something with this in event_route[dialog:failed], does it still error? Fred Posner The Palner Group, Inc. f...@palner.com @fredposner Good. Fast. Cheap. <- pick two ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Adding dialogs to a profile in a failure_route
On 04/28/2014 06:54 PM, Alex Balashov wrote: I don't think that will work, because no dialog is created by the 302 redirect. 11.3. event_route[dialog:failed] Executed when dialog is not completed (+300 reply to INVITE). --fred ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Adding dialogs to a profile in a failure_route
On 04/30/2014 04:16 AM, Alex Balashov wrote: So, do you suppose I could achieve my objective by tracking every dialog--that is, by calling dlg_manage() in the initial request route for every call, and then calling set_dlg_profile() out of a failure route conditionally? Wouldn't you need to set it within the original invite and then "do something" with it in the event_route[dialog:failed]? Fred Posner The Palner Group, Inc. 503-914-0999 (direct) 954-472-2896 (fax) Good. Fast. Cheap. <- Pick two. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] avpops issue
For the is_user_in... are you loading the group module? For avp_write, that function hasn't existed in some time. You can use logic such as: $avp(s:fwd_blind) = $ru; Fred Posner The Palner Group, Inc. 503-914-0999 (direct) 954-472-2896 (fax) Good. Fast. Cheap. <- Pick two. On 05/22/2014 04:49 PM, Gilbert T. Gutierrez, Jr. wrote: > I am trying to setup call forwarding but I am getting a failure when I > attempt to implement it. > > Kamailio Version 4.1.1 > Centos 6 x64 > Using precompiled RPMs from telephony.repo > > I am following guidance from the following urls... > http://www.kamailio.org/wiki/tutorials/mini-howto-admin/call_forwarding > http://www.kamailio.org/dokuwiki/doku.php/tutorials:avpops > http://www.kamailio.org/dokuwiki/doku.php/examples:set-blind-call-forwarding > > > > 0(16018) ERROR: [cfg.y:3272]: yyparse(): cfg. parser: failed to > find command is_user_in > 0(16018) : [cfg.y:3411]: yyerror_at(): parse error in config > file //etc/kamailio/kamailio.cfg, line 1046, column 41: unknown command, > missing loadmodule? > > 0(16018) ERROR: [cfg.y:3272]: yyparse(): cfg. parser: failed to > find command avp_write > 0(16018) : [cfg.y:3411]: yyerror_at(): parse error in config > file //etc/kamailio/kamailio.cfg, line 1048, column 39: unknown command, > missing loadmodule? > > My understanding is that when I load module avpops that it should > include those 2 procedures (avp_write and is_user_in). I am loading the > module avpops.so > > Can someone please set me straight. Thank you. > > Gilbert T. Gutierrez, Jr. > Phoenix Internet > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls
do you have an ngrep of the sip traffic? This can happen if the sip/rtp cannot connect (perhaps blocked by the dsl router) Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 07/01/2014 01:12 PM, Carlos Rangel wrote: > Hello List > > > > Hopefully someone can help. This is the problem when the call is hug up 20-30 > seconds after it initiates. The call is only hung on when the remote > extension initiates the call. If the remote extension receives the call there > is no problem the call is not hung on. I changed the remote cisco phone for a > yealink and it is the same behavior. It thought it was the phone. > > > > This is what I am using in kamailio.cfg > > > > #!define WITH_MYSQL > > #!define WITH_AUTH > > #!define WITH_ASTERISK > > #!define WITH_USRLOCDB > > #!define WITH_ANTIFLOOD > > > > Remote User Internet > Internal network > > Yealink IP TG28P DSL router ---|--Internet |-Cisco ASA > 5500 FW--Kamailio/Freepbx (Same Box)--IAX > Trunk--Freepbx Production Server |-- PSTN > > > > > > Thanks > > Carlos Rangel > > > > De: Carlos Rangel [mailto:cran...@globaltelesourcing.com] > Enviado el: jueves, 26 de junio de 2014 01:27 p.m. > Para: mico...@gmail.com; 'Kamailio (SER) - Users Mailing List' > Asunto: RE: [SR-Users] Kamailio Freepbx Integration Dropping Calls > > > > Hi Daniel > > > > Thank you so much for your response. Here is the SIP trace of one of the > calls, I am not sure where the call initiates but you can see at the end of > the file in bold X-Asterisk-HangupCause: No user responding. I am not sure > why is it sending this message though. > > > > The variables are > > > > Extension/Username=X > > Ext_IP= Public IP > > Internal_IP= Asterisk/Kamailio internal IP > > > > Sorry for the long file but again I am not sure where the call initiates > > > > This is the part where that call is hung on. > > > > U 2014/06/26 13:36:11.831965 Kamailio_IP:5080 -> Kamailio_IP:5060 > > BYE sip:X@65.190.71.203:5060;user=phone;transport=udp SIP/2.0. > > Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47. > > Route: . > > Max-Forwards: 70. > > From: ;tag=as2e670ea4. > > To: "User" ;tag=000653dc394000970f227678-1fafb4e2. > > Call-ID: 000653dc-394b-33caf1b2-20ccd185@192.168.0.22. > > CSeq: 102 BYE. > > User-Agent: FPBX-2.11.0(11.10.2). > > X-Asterisk-HangupCause: No user responding. > > X-Asterisk-HangupCauseCode: 18. > > Content-Length: 0. > > . > > U 2014/06/26 13:36:11.832260 Kamailio_IP:5060 -> 65.190.71.203:5060 > > BYE sip:X@65.190.71.203:5060;user=phone;transport=udp SIP/2.0. > > Via: SIP/2.0/UDP > Kamailio_IP;branch=z9hG4bKcf68.d6ef5aa9cc5bd0fb0ab13a563b7cf284.0. > > Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47. > > Max-Forwards: 69. > > From: ;tag=as2e670ea4. > > To: "User" ;tag=000653dc394000970f227678-1fafb4e2. > > Call-ID: 000653dc-394b-33caf1b2-20ccd185@192.168.0.22. > > CSeq: 102 BYE. > > User-Agent: FPBX-2.11.0(11.10.2). > > X-Asterisk-HangupCause: No user responding. > > X-Asterisk-HangupCauseCode: 18. > > Content-Length: 0. > > > > > > > > > > Descripción: Description: Description: DLR-Logo-No-TextCARLOS RANGEL | > INFORMATION TECHNOLOGY DIRECTOR > > Global Telesourcing México, S. de R.L. de C.V. | Aarón Sáenz #1891-1 | > Monterrey, N.L., México > Direct 703 894 1667 | Mobile US 703 894 1667 | Mobile MX +52 1 812 000 7362 | > cran...@globaltelesourcing.com > > > > The information contained in this e-mail and any attached documents may > contain information that is confidential or otherwise protected from > disclosure. If you are not the intended recipient of this message, or if this > message has been sent to you in error, please immediately alert the sender by > reply e-mail and then delete this message, including any attachments. Any > dissemination, distribution or other use of the contents of this message by > anyone other than the intended recipient is strictly prohibited. > > > > > > De: sr-users-boun...@lists.sip-router.org > [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Daniel-Co
Re: [SR-Users] Calls per second
I've done this with dialog and a sql lookup/update. The sql call updates the table with how many calls are in total for the "client" and how many international calls. There's a max calls and a max international. If current < max, the call can go through. The db allows me to combine the lookup over multiple media servers and kamailio servers. The lookup checks the db so any modifications occur in real-time. Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 07/03/2014 07:37 AM, Olle E. Johansson wrote: > Hi! > > Have you implemented a per-customer rate limit in Calls per second? If so - > how? > > I've played with ratelimit/pipelimit and it seems like I can define a > database with one pipe per customer - but have to restart Kamailio to add > customers. There are warnings for low timer settings, like 1 second, but I > don't know how up-to-date those warnings are. > > I guess I could play with hash tables and implement something pike-like > there, but it seems like a workaround for something pretty common. > > So the question remains - how are you limiting on a cps per customer? > > /O > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Calls per second
I think my head isn't fully woken up yet -- sorry about that. Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 07/03/2014 07:48 AM, Olle E. Johansson wrote: > I am looking for calls setups per second - not concurrent calls. > Sorry for not being exact. > > /O > > On 03 Jul 2014, at 13:41, Fred Posner wrote: > >> I've done this with dialog and a sql lookup/update. >> >> The sql call updates the table with how many calls are in total for the >> "client" and how many international calls. There's a max calls and a max >> international. >> >> If current < max, the call can go through. >> >> The db allows me to combine the lookup over multiple media servers and >> kamailio servers. The lookup checks the db so any modifications occur in >> real-time. >> >> Fred Posner >> The Palner Group, Inc. >> http://www.palner.com (web) >> +1-503-914-0999 (direct) >> +1-954-472-2896 (fax) >> >> On 07/03/2014 07:37 AM, Olle E. Johansson wrote: >>> Hi! >>> >>> Have you implemented a per-customer rate limit in Calls per second? If so - >>> how? >>> >>> I've played with ratelimit/pipelimit and it seems like I can define a >>> database with one pipe per customer - but have to restart Kamailio to add >>> customers. There are warnings for low timer settings, like 1 second, but I >>> don't know how up-to-date those warnings are. >>> >>> I guess I could play with hash tables and implement something pike-like >>> there, but it seems like a workaround for something pretty common. >>> >>> So the question remains - how are you limiting on a cps per customer? >>> >>> /O >>> ___ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> ___ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio behind NAT- alias issue
Hello Yuriy, > If I write at kamailio.cfg: > alias=sip.myserver.com > > I see error at log - bad_uri sip.myserver.com try adding the port... alias=sip.myserver.com:5060 Also, since you're behind NAT make sure you also advertise the address with advertised_address="sip.myserver.com". Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 08/02/2014 07:51 AM, Yuriy Gorlichenko wrote: > Hello. I have Kamailio running behind NAT. It lesten eth0 with ip > 192.168.0.3 and > I have external IP that have domain name (for example sip.myserver.com). > > Register packets from clients comes from external IP. > > If I write at kamailio.cfg: > alias=sip.myserver.com > > I see error at log - bad_uri sip.myserver.com > > I think it happens because kamailio does not know atything about external > ip pecause kamamiliol working with server interfaces (eth0, eth1, etc.) > > So my question - how to listen external IP with domain name on kamailio > that running behind NAT? > > P.S. I do not have any access to router, that present NAT for me. > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Asterisk cluster behind kamailio natted to pubic IP, presenting internal ip addresses in From tag
On 08/28/2014 12:44 PM, Tim Chubb wrote: > Hi All > > ...snip... > > So far everything is working fine, I can register via the public IP > address, IM & presence is working, and as does audio, however when I > dial an extension the caller id comes up like this > 12345@172.16.15.123:5080 <mailto:12345@172.16.15.123:5080> which is the > ip addresses of the asterisk server that the dispatcher has assigned to > the call, I have tried setting the P-Asserted-Identity, > P-Preferred-Identity & Remote-Party-ID headers to no effect. What I > would like to achieve is that the public IP or domain name comes up when > I call an extension. From examining the sip traffic traversing the > kamailio box, it seems that the From and Contact headers sent by > asterisk are the source of the internal implementation information > reaching the end-user. > > .../snip... > > > *Tim.* > I'm assuming with the 5080 that this call goes through the Asterisk box before hitting the registered user on Kamailio... if that's correct, have you also forced a CALLERID(name) on the call? A grep of the sip traffic would show if you have something perhaps removing this information before sending to the client. Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] paid support plans
On Tue, 2014-09-16 at 19:46 +, Mike Hancock wrote: > Hello, > > [SNIP] I would like to purchase support to help to get this up and > running as well as configured to get us going. [/SNIP] > > > Mike Hancock Hi Mike, This is the link for the business directory: http://www.kamailio.org/w/business-directory/ "Disclaimer: whilst we do the best to select serious applications, Kamailio and SER projects, along with their developers and management groups, do not guarantee nor take responsibility for the quality of services or products offered by the companies listed here. It is your sole decision to do business with any of the entities listed here and all commercial relations and liabilities are only between you and your business partner, without any involvement of the two open source projects." On that page, you can find a great company to help you. --fred ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX
Hi Kenny, This depends on the carriers and scenarios that you may use. I know "depends" is a horrible answer, but one of the great aspects of Kamailio is the flexibility of the modules. Some deployments may have a group of Asterisk servers all configured similarly for handling calls. With this type of scenario, you would benefit from using the dispatcher module. Many people like to use Kamailio on the public side of their network and keep their asterisk servers on the private. This would be an example of when to use rtpproxy (in bridge mode). Some carriers hate seeing the chain of systems on your network (ie the asterisk boxes). Sometimes the use of TOPOH helps to integrate with the carriers who have chosen their own "interpretations" of RFC for "security." And there's more... The bottom line, is that the devil is in the details. Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 10/23/2014 09:12 AM, Kenny Watson wrote: Hi, I have a few asterisk servers providing some basic SIP trunking and routing. We have remote PBXs trunked onto asterisk which calls come into asterisk and are routing down to extensions on the remote PBX via prefix routing. I’m looking to have a central Kamailio Registrar/Proxy/Loadbalancer which Invites come into and are routed out to either SIP phones which are registered or to the remote PBX. I’m looking for some advice as to which modules would be best to use to achieve this as the remote PBXs will be dynamically registered rather than fixed gateways. Please let me know what further information would be helpful. Thanks Kenny Watson ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX
If you want to call a user on Kamailio from Asterisk... example... exten => s,1,Verbose(4,calling user on kamailio) same => n,Dial(SIP/USERNAME@KAMAILIO,time,options) same => n,--after dial logic -- Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 10/23/2014 11:21 AM, Kenny Watson wrote: Hi Fred, Thanks for the quick response. I already do use some Kamailio features on our internal network for load balancing. The use case that I'm interested in is to effectively replace an asterisk server that I use for SIP trunking to remote phone systems with a Kamailio registrar/proxy and a bank of asterisk servers placing calls direct to extensions on the remote PBX. I currently have this running on asterisk which I route to the different remote PBX extensions using prefix based routing down to the destination peer on asterisk which is essentially what I need to replicate on Kamailio. i.e. 2021 routes to @remotepbx1 remotepbx1 maybe defined as either by IP address or via a "normal" registered sip peer with a username/password combo. I understand that I can dial a registered device directly but its how to call a remote extension on a registered device via Kamailio. Thanks Kenny Watson -Original Message- From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Fred Posner Sent: 23 October 2014 16:00 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Kamailio Infront of Asterisk with remote PBX Hi Kenny, This depends on the carriers and scenarios that you may use. I know "depends" is a horrible answer, but one of the great aspects of Kamailio is the flexibility of the modules. Some deployments may have a group of Asterisk servers all configured similarly for handling calls. With this type of scenario, you would benefit from using the dispatcher module. Many people like to use Kamailio on the public side of their network and keep their asterisk servers on the private. This would be an example of when to use rtpproxy (in bridge mode). Some carriers hate seeing the chain of systems on your network (ie the asterisk boxes). Sometimes the use of TOPOH helps to integrate with the carriers who have chosen their own "interpretations" of RFC for "security." And there's more... The bottom line, is that the devil is in the details. Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 10/23/2014 09:12 AM, Kenny Watson wrote: Hi, I have a few asterisk servers providing some basic SIP trunking and routing. We have remote PBXs trunked onto asterisk which calls come into asterisk and are routing down to extensions on the remote PBX via prefix routing. I'm looking to have a central Kamailio Registrar/Proxy/Loadbalancer which Invites come into and are routed out to either SIP phones which are registered or to the remote PBX. I'm looking for some advice as to which modules would be best to use to achieve this as the remote PBXs will be dynamically registered rather than fixed gateways. Please let me know what further information would be helpful. Thanks Kenny Watson ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Failed to install Kamailio database
Do you have mysql installed? Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 10/24/2014 08:52 PM, Mahmoud Ramadan Ali wrote: Hiii everyone, I can not create kamailio database and get this error message... any ideas ? Thanks in advance... ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Failed to install Kamailio database
You will need to install mysql if you would like to use a mysql database. It is not required that you use mysql. Other databases are supported as well as a database not being a requirement for the software. Fred Posner On 10/24/2014 08:57 PM, Mahmoud Ramadan Ali wrote: No ! i do not have mysql installed...does the script will install it for me or i should install it previously ? On 10/24/14, Fred Posner wrote: Do you have mysql installed? Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 10/24/2014 08:52 PM, Mahmoud Ramadan Ali wrote: Hiii everyone, I can not create kamailio database and get this error message... any ideas ? Thanks in advance... ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] RFC: infrastructure upgrade - git, tracker, ...
I'm certain that LOD would be willing to sponsor the server for git / tracker and I'd offer to handle the sysadmin of the server. Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 11/05/2014 09:25 AM, Daniel-Constantin Mierla wrote: Hello, as most of you know, we have a rather distributed infrastructure, with servers provided from different companies or persons. We came to the time when one of the servers is too old and considered to be decommisioned, so we have to decide how to move on. It is about sip-router.org, who was offered/sponsored by courtesy of Jan Janak. From project point of view, the server hosts: - git repository - bug tracker - website and wiki under domain sip-router.org Given that no matter what we like, there is work to do, I am looking to see what are the best options out there for everyone involved in the project. The sip-router.or wbesite and wiki, which are not really updated anyhow, will be relocated as virtual host in kamailio.org server and made static for historic purposes. For git and tracker, I thought of two variants: 1) move to use an external hosting service - the first candidate and perhaps the only to be considered is GitHub, we have there already a real time mirror of git repository. Then we should get a read only mirror to kamailio.org. If the tracker on github is good enough for everyone, then we will use it. I could see quickly that lot of kamailio developers already have an account on github. 2) get a new server and relocate those components there. It will need to be configured from scratch and the components eventually updated to use latest versions. In case of tracker, we have eventually to re-evaluate if flyspray worth keeping, as we had several discussions, due to the fact that the project doesn't seem to be very active. My personal preference at this moment is 1), given that offloads administration works from project. For 2) we will need someone to commit to a sysadmin job for long time. As probably you noticed lately, serious security vulnerabilities can appear and someone needs to take care of proper maintenance of the server. I don't want to get all services on kamailio.org, as it has other critical components (mailing lists, releases, website, ...) and messing it or overloading doesn't make sense. There is no real pressure to come to a decision, we can still rely on the server for a while, but I would rather not postpone it for long. While users are encouraged to give their opinion, I feel that existing developers should have the main role in decision, being something that impacts them directly. Your preference? Any other opinions? Cheers, Daniel ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] RFC: infrastructure upgrade - git, tracker, ...
With no private repositories, gitlab is free which is nice. ---Fred > On Nov 9, 2014, at 1:20 PM, Jan Janak wrote: > >> On Thu, Nov 6, 2014 at 11:40 AM, Jan Janak wrote: >> If you prefer to keep a self-hosted git repository, I think we should at >> least move it to gitolite: >> http://gitolite.com/ >> >> To make it more manageable (doesn't require ssh access for individual >> people). > > Gitlab is another potential candidate for a self-hosted repository: > > https://about.gitlab.com/ > > (seems to provide more than gitolite). > > -Jan > >> >>> On Wed, Nov 5, 2014 at 9:25 AM, Daniel-Constantin Mierla >>> wrote: >>> Hello, >>> >>> as most of you know, we have a rather distributed infrastructure, with >>> servers provided from different companies or persons. >>> >>> We came to the time when one of the servers is too old and considered to >>> be decommisioned, so we have to decide how to move on. It is about >>> sip-router.org, who was offered/sponsored by courtesy of Jan Janak. From >>> project point of view, the server hosts: >>> >>> - git repository >>> - bug tracker >>> - website and wiki under domain sip-router.org >>> >>> Given that no matter what we like, there is work to do, I am looking to >>> see what are the best options out there for everyone involved in the >>> project. >>> >>> The sip-router.or wbesite and wiki, which are not really updated anyhow, >>> will be relocated as virtual host in kamailio.org server and made static >>> for historic purposes. >>> >>> For git and tracker, I thought of two variants: >>> >>> 1) move to use an external hosting service - the first candidate and >>> perhaps the only to be considered is GitHub, we have there already a >>> real time mirror of git repository. Then we should get a read only >>> mirror to kamailio.org. If the tracker on github is good enough for >>> everyone, then we will use it. I could see quickly that lot of kamailio >>> developers already have an account on github. >>> >>> 2) get a new server and relocate those components there. It will need to >>> be configured from scratch and the components eventually updated to use >>> latest versions. In case of tracker, we have eventually to re-evaluate >>> if flyspray worth keeping, as we had several discussions, due to the >>> fact that the project doesn't seem to be very active. >>> >>> My personal preference at this moment is 1), given that offloads >>> administration works from project. >>> >>> For 2) we will need someone to commit to a sysadmin job for long time. >>> As probably you noticed lately, serious security vulnerabilities can >>> appear and someone needs to take care of proper maintenance of the >>> server. I don't want to get all services on kamailio.org, as it has >>> other critical components (mailing lists, releases, website, ...) and >>> messing it or overloading doesn't make sense. >>> >>> There is no real pressure to come to a decision, we can still rely on >>> the server for a while, but I would rather not postpone it for long. >>> >>> While users are encouraged to give their opinion, I feel that existing >>> developers should have the main role in decision, being something that >>> impacts them directly. >>> >>> Your preference? Any other opinions? >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla >>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com >>> >>> >>> ___ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
