e 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
t; 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 hin
$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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r 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
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 eith
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 an
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
x 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
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
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 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 t
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 rega
gards,
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
>
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
n 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
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
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 Cen
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 {
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';
>>
>>
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
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,
F
s 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
___
led
> > 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://k
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-300-1
es) 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
___
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
g 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
__
best
way I can contribute to the project; refund administrator.
--
fred
http://qxork.com
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e 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
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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
u 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. Ov
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
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
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
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
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 Ex
/ 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
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sr-
rvers? 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
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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
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
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
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
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
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
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 As
or 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)
___
S
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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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
pected.
ERROR: rtpengine [rtpengine.c:1622]: rtpp_test(): proxy responded with invalid
response
Any advise would be nice.
Thank you
Fred
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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
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
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
cally 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
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rnal 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, In
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
___
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ort() 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&q
ds... it just continues on the mailing lists. =)
Best regards,
Fred Posner
http://www.palner.com
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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 :)
&
ould 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
scenari
ooks/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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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
> s
ibutes 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 fla
on 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 a
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 Asteri
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, 2600
me to use additional
authentication methods.
I believe Polycom still max's out at 32.
--fred
0x42AE1A40.asc
Description: application/pgp-keys
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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:
>>>
>>
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
> 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
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:
>
>
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
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
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
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
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
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 (
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 rang
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.
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
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 s
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", &quo
> 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
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 $sq
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
f 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)
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 Ex
ed 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
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 05/22/20
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 L
e 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 wro
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.
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="si
c 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 (Open
es 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 gre
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.
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 (d
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
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
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 Mi
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
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