Maybe a restart on kamailio service wasn't done. Or maybe you're editing a
different config file while kamailio is using something else !
Otherwise I dont see anything missing.
--
Sammy
On Mon, Apr 10, 2017 at 4:08 PM, Nicolas Pace wrote:
> Hi everyone,
>
> I'm having some trouble setting up a
Hi,
Interesting to know that AWS has PPS limitation.
Found this article to be useful :
http://techblog.cloudperf.net/2016/05/2-million-packets-per-second-on-public.html
Regards,
Sammy
On Tue, Mar 28, 2017 at 3:37 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> besides Alex' suggestion to loo
ialog" bug
>
> modparam("pua", "update_period", 20)
>
> modparam("pua", "outbound_proxy", MY_SIP_URL)
>
>
>
> modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO)
>
> modparam("pua_dialoginfo",
g' indicates
> only different call states, but also hold more info about a call. In
> 'dialog' case XML does not even has an option to indicate, that phone is
> online or offline.
>
> Refer to *https://tools.ietf.org/html/rfc3856*
> <https://tools.ietf.org/html/rfc3856> and
>
dialog notifications is not possible,
> not in the specs, but eventually you can write a module yourself and map as
> you want/need presence states over dialog states.
>
> Cheers,
> Daniel
> On 03/03/2017 19:13, SamyGo wrote:
>
> Hi,
> I'm in need of making/tweak an e
Hi,
I'm in need of making/tweak an existing Kamailio Presence setup which is
giving some tough time.
*Whats already working:*
BLF dialog states changes are already sent across the users. SCA is working
as well.
*What isn't working:*
When a User comes online then it sends SUBSCRIBE with *Event: di
Hi Again,
You need to enable NAT handling in your Kamailio (#!define WITH_NAT), then
depending upon how your clients will interact with asterisk you may or may
not need a media proxy, like RTPproxy. If asterisks can send/receive media
directly from the internet then its ok for now, else you defini
.github.com/ticklingcontest/0b46e8e53bf50aa3875395c8fb86ff66
>
> BTW, I moved away from dispatcher.list file as I could not get it to work
> (the DB values always registered, but the list file didn't take).
>
> Any insight is appreciated.
>
> Thanks!
>
>
>
>
;1", "4") evaluating to false for me?
>>>
>>> My dispatcher info:
>>>
>>> root@kamailio:/etc/kamailio# kamctl dispatcher show
>>> dispatcher gateways
>>>
>>> ++---+---+---+--+---+--
Hi Aqs,
How about putting a loop protection of some sort additionally like $si !=
myself.
Is there any chance the API http call is getting skipped to relay w/o
getting any reply back and hence looping back to itself ?
Regards,
Sammy
On Jul 22, 2016 14:17, "Aqs Younas" wrote:
Hi,
I am using htt
Hi Tickles,
a) Have you tried doing this on Asterisk realtime ? for any regular direct
registering extension w/o kamailio(even before kamailio) ? If an extension
registers to one asterisk the rest of the boxes would know where to contact
this extension via realtime-db ? I'd say give that a try and
Hi Marino,
How do you mean execute a route !? like enable a route for all calls
onwards after the RPC command execution ? or for a specific call engage
that route ?
If this is to enable/disable a particular route for ALL calls based on a
RPC command I'd make use of some DB or memcache variable to
ng/vrf.txt
>
> Regards.
>
> Le 12 juil. 2016 19:45, "SamyGo" a écrit :
>
>> Hi,
>>
>> Question:
>> Is there a function in kamailio to get the interface/listen IP kamailio
>> would use to route a packet out for a domain / ip ?
>>
>>
Hi,
Question:
Is there a function in kamailio to get the interface/listen IP kamailio
would use to route a packet out for a domain / ip ?
Example:
$avp(interface_ip) = find_interface_for_request("192.168.12.123");
And it gives me the Kamailio's Listen address that would be used for
sending call
>
> maybe you can loop through a 2nd instance of kamailio (can be same server,
> different port) that has topoh enabled.
>
> Otherwise, if the gateway is matching on full r-uri, you can try to add
> some extra uri params, which are not relevant for target number.
>
> Cheers,
&
Hi,
I've a very strange scenario to work on which requires me to parallel fork
the call to the same Destination provider. The only problem here is that
they think that the second INVITE with different branch tag is a
re-transmission and hence only take one call forward.
I do not have to modify an
Hi Collin,
I can only think that by doing this saving of nonce, and accessible by rest
of the boxes in cluster, isnt it going to put the authentication mechsnism
at risk ? Even if not, that means all the servers in your cluster supposed
to behave predictably same ! Hence again security concern !
Can you do the same with either reids on DB=1 or remove db=1 string from
the ndb_redis url.
redis-cli
127.0.0.1:6379> select 1
127.0.0.1:6379[1]> GHET usuario:2001 status
OR
*modparam("ndb_redis", "server", "name=srvN;addr=127.0.0.1;port=6379")*
If that doesn't help I'll be happy to share the r
Hi,
I dont have my config at cell will share mine later. Meanwhile can you try
HGET only to pull that 1 value plus use
$redis(res=>value) with that.
Regards,
Sammy
On Jun 28, 2016 02:25, "Fabian Pignataro"
wrote:
> Hello Community,
>
> I'm trying to read data (hash) stored in redis database, from
Hi,
What kind of design you've in mind when thinking about BIN interface ? What
are the things that you want to replicate ? Just call dialog states ?
Given some more usage clarification, someone might be able to guide further
on what can be used.
Regards,
Sammy
On Sat, Jun 18, 2016 at 2:15 AM,
Hi Alexandru,
The Expire header in Register can be your clue here. if it is greater than
0 means its a registration attempt. If the value is 0 then it is
un-register request.
This is how you can catch this.
if(is_method("REGISTER") ) {
if($hdr(Expire) == 0) {
// Un-Registering User $fU
} els
Hi,
Since DBs for both are replicated and I assume user location table is
teplicated as well. In this case yes the secondary server will be able to
detect and find the called number as online.
The only exception would be the received socket. You might have to force
local socket to send INVITE to c
redis is capable for sure of handling that size of records. If you just
> keep (key, value) pairs, redis is probably the best to choose because is
> know to be very fast for looking up on a key.
>
> Cheers,
> Daniel
>
> On 12/05/16 22:54, SamyGo wrote:
>
> Hi Daniel,
&
re exposed plans to eventually update it.
>
> Are you using the old cassandra lib version?
>
> Cheers,
> Daniel
> On 12/05/16 18:17, SamyGo wrote:
>
> Hi List,
>
> I'm trying to hook up userblacklist module with db_cassandra. I've
> kamailio keyspace configured
hould have some operations allowing matching by key or prefix.
>
> Cheers,
> Daniel
>
>
> On 11/05/16 15:22, SamyGo wrote:
>
> Hi,
>
> I am tasked to make use of blacklist module for about 4 to 6 million
> numbers. I am thinking of using Cassandra for the purpose
Hi List,
I'm trying to hook up userblacklist module with db_cassandra. I've kamailio
keyspace configured with tables for userblacklist , globalblacklist, and
version are created with some data in there.
Here is setup info:
version: kamailio 4.4.1
*kamailio.cfg*
...
loadmodule "db_cassandra.so"
Hi,
I am tasked to make use of blacklist module for about 4 to 6 million
numbers. I am thinking of using Cassandra for the purpose but reading
through the documentatiom of module and recent mailing list discussion made
me a bit hesitant.
I am looking for advise on this whether this is going to pe
Hi Fuad,
I think you should put the db return value in am avp before using it in the
sl_send_reply.
$avp(ip) = $avp(ra=>ip)
Or whatever the syntax is. Make sure that the DB query is returning the
required value.
Also I guess just using a 302 is not enough and you need to set the proper
contact hea
Hi Igor,
If Im understanding this correctly you have a separate Kamailio just to
handle Presence.
In this case you mention that a SUBscribe has been acknowledged by this
Presence Kamailio but it does not do anything further!!
What are the chances that this new Presence kamailio is separate from the
I just took a printout of it and will be framing it in my office. :D
My vote for Alex B.
On Apr 1, 2016 07:27, "Alex Balashov" wrote:
> For immediate release:
>
> ATLANTA, GA (1 April 2016)--Alex J. Balashov, a self-styled
> businessman based in Atlanta, Georgia, USA, has a plan to "Make
> Kamai
Yes
On Sat, Mar 26, 2016 at 2:47 AM, Barış Şekerciler <
baris.sekerci...@outlook.com> wrote:
> Hello Sammy,
> Thanks your for reply.
>
> I could not find the */root/.pgpass. *Both Kamailio and PGS server.
> Should I manually create it?
>
> Regards,
> Barış.
>
> --
> Da
Hi,
I've experienced the same in past couple of weeks, apparently if you've the
file */root/.pgpass* with the pgsql user/password in it something like
*:*:*:kamailio:kamailiopass
Then it should not ask you over and over again. I kept pasting password
until it was done doing it's DB stuff, so I ca
Hi Brooks,
While I was trying to figure issue out with my dispatcher issue here is
something that I think might work for you. It do requires modifying little
bit C code to pull some weight from a redis/memcache of remote destination
and use that value instead. Somewhere in this function of dispatc
uture reference, For algo 9 always sum up weights
for destinations to equal 100.
Thanks again for reading, and replying Brooks.
Regards,
Sammy
On Thu, Mar 24, 2016 at 2:08 PM, SamyGo wrote:
> Hi Brooks,
>
> Well no I'm using algo 9, with weights assigned to each IP in the set.
&
cations*
>
> 5190 Golden Foothill Pkwy
>
> El Dorado Hills, CA 95762
>
> *office:* 916.235.2097 | *main:* 888.444., Option 2
>
> *email:* bbrid...@o1.com | *web:* www.o1.com
>
>
>
> *From:* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
> Behalf Of
Hi All,
I'm having a wee little bit of difficulty in trying to load-balance
registrations to multiple servers based on their weights.
Kind of similar scenario as discussed in this thread:
http://lists.sip-router.org/pipermail/sr-users/2014-December/086235.html
The problem I'm facing is that disp
Hi,
Well reading might be a very good idea but trying things and scenarios will
be an even better thing.
There are very amazing blogs, more like How to for kamailio written by
Daniel himself available over asipto knowledge base. Kb.asipto.com
Just go through those installation steps, copy over conf
Hello Dadang,
...inside route[LOCATION]
*Server1:*
if(!lookup("location") {
xlog("Failed to Find user online in this server, maybe try Second
server\n")
$ru = "sip:" + $rU + "@192.168.15.30";
route(RELAY);
}
*Server2:*
if(!lookup("location") {
xlog("Failed to Find user o
Hi Prashant,
If I were to do it the lazy way and if I have understood your requirement
completely. I might've ended up using snmp monitoring of the CPU and set a
memcache/redis variable to be 0 or 1 in case CPU goes above or below 70.
In my kamailio.cfg I would just check that redis variable and de
Hi,
I think the best guide closest to your description is here :
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
Here is what you need to do. (*Besides mentioning what you tried and what
problems were faced*).
1 - Configure kamailio to use the DB schema where your use
value of $ru with xlog and see if it has username?
>>
>> Cheers,
>> Daniel
>>
>>
>> On 16/02/16 22:54, Igor Olhovskiy wrote:
>>
>> Yep, got same problem.
>> Not sure it’s a bug, but seems to be topoh module is really lack some
>> configuration.
Hi,
I don't think that the cluster module is recognized outside the
kamailio.cfg - In order to get your two Databases to work with DBHOST
string is maybe get a MySQL proxy used between the kamctl and the two
Databases. The MySQL proxy would decide if the first DB is down hence
connect to secondary
Fri, Feb 12, 2016 at 2:33 PM, malik sherif wrote:
> Thanks Sammy, I will use pastebin.com next as you recommended.
>
> Thanks
>
> Abdul
>
>
> --
> *From:* sr-users on behalf of
> SamyGo
> *Sent:* Thursday, February 11, 2016 11:50
Hi All,
I've recently stumbled upon this little hitch while using kamailio with
topoh module that the Contact header do not contains the User part for the
302 Moved temporarily packet.
*My topology:*
UserA<\
->Kamailio<===>FreeSwitch
UserB
DESTROY
> 2016-02-11 11:38:42.669308 [DEBUG] mod_sofia.c:323 sofia/internal/
> 1...@newkama.abdulkamailiosip.com SOFIA DESTROY
> 2016-02-11 11:38:42.669308 [DEBUG] switch_core_state_machine.c:111
> sofia/internal/1...@newkama.abdulkamailiosip.com Standard DESTROY
> 2016-02-11 11:38:42.66
on kamailio site.
>
> Thanks
>
> Abdul
>
>
> ------
> *From:* sr-users on behalf of
> SamyGo
> *Sent:* Wednesday, February 10, 2016 10:23 PM
>
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] Fw: Kamailio and freeswitch integration for SBC
>
>
se
> kamailio+rtpengine to decrypt/encrypt the leg to webrtc and get sip and
> plain rtp to this transcoder.
>
> Cheers,
> Daniel
>
>
> On 10/02/16 22:12, SamyGo wrote:
>
> Thanks for clarification Daniel. That obviously mean that I can not
> achieve transc
; From Freeswitch dial plan
>
>
>
>
>
>
>
>data="hangup_after_bridge=true"/>
> data="sip_invite_domain=AbdulkamailioSIP.com"/>
>
ltiplexing streams).
>
> Cheers,
> Daniel
>
> On 10/02/16 20:49, SamyGo wrote:
>
> Hi All,
>
> reference to this link:
> https://www.kamailio.org/wiki/devel/rtcweb_breaker#scenarios
>
> I want to know if the module to communicate with RTCWeb Breaker is
> avai
Hi All,
reference to this link:
https://www.kamailio.org/wiki/devel/rtcweb_breaker#scenarios
I want to know if the module to communicate with RTCWeb Breaker is
available or it was just a proposal and no more under consideration.
I have webrtc clients registered to Kamailio but due to lack of
(sc
Hi,
Interesting discussion going on. Ive to ask can OP use shared location
table between different kamailio servers and use db mode 3.
All he has to do is if UA B calls UA A via Kamailio2 he just needs to find
the received socket and if it is not local Kamailio2 then route to the IP
maybe attach a
Hi Rehan,
No matter which mode you are running rtpproxy in that IP will always be the
IP of the machine it is running on.
That means that SDP will take that IP once routed to locally subnet A2B
servers.
As far as the A2B detecting SIP user as online or offline based on DB, I
am not too sure about
ders in sky blue, that doesn't mean there is any error in
there.
Regards,
Sammy
On Fri, Jan 29, 2016 at 11:47 AM, SamyGo wrote:
> Hi Abdul,
>
> This is where you are getting your 404 NOT Found from Kamailio:
>
>
>
> On Thu, Jan 28, 2016 at 4:30 PM, malik sherif
> wr
Hi Abdul,
This is where you are getting your 404 NOT Found from Kamailio:
On Thu, Jan 28, 2016 at 4:30 PM, malik sherif wrote:
> I will also run the commands that suggested.
>
>
> --
> *From:* sr-users on behalf of
> SamyGo
> *Sent:* Thursda
I believe Daniel is busy with FOSDEM ,
Abdul can you confirm that you're still getting this output in FS console:
2016-01-13 05:37:29.572184 [INFO] mod_dialplan_xml.c:635 Processing
7632689991 <7632689991>->kb-7632689993 in context default
2016-01-13 05:37:29.572184 [CRIT] mod_dptools.c:1638 WAR
dio 10076 RTP/AVP 97 98
> c=IN IP4 rtpproxy1.example.com
> a=rtpmap:97 speex/8000
> a=fmtp:97 vbr=on
> a=rtpmap:98 telephone-event/8000
> m=video 20007 RTP/AVP 96
> c=IN IP4 rtpproxy2.example.com
> a=rtpmap:96 VP8/9
>
>
>
> On Thu, Jan 14, 2016 at 8:31 PM, SamyG
his case, most probably video and audio rtp packages will flow through
> rtpproxy-2.
> However what I want to do is to keep rtpproxy-1 for audio packages and add
> rtpproxy-2 for video packages
> and release audio ports from rtpproxy-1 and video ports from rtpproxy-2
> when BYE
Hi,
Yes thats possible as I think you can do a search in SDP body for "video"
if found then select the rtpproxy instance else select other one.
Regards,
Sammy
On Jan 14, 2016 06:39, "Koray Vatansever"
wrote:
> Hi everybody,
>
> Is there a way to use different rtpproxies for different media types
It is possible. I created a blog entry from a mailing list thread doing
this sort of stuff for Asterisk behind Kamailio and voicemail MWI .
http://saevolgo.blogspot.ca/2012/07/asterisk-behind-kamailio-voicemail-mwi.html?m=1
The scripts and everything should just give you idea. In Kamailio when you
Hi,
You can catch this NOTIFY in kamailio and execute a script using sipsak
like perl script doing some db/file/sipsak stuff.
You might find examples of such thing in few older threads.
On Jan 2, 2016 18:55, "Arsen Hovhanissian" wrote:
> Hi everyone, I’m trying to send a NOTIFY event using “sipsa
Hi,
I believe you're trying to setup some MOH-Queue it can't find the MOH file.
0(4420) ERROR: mohqueue [mohq_db.c:504]: update_mohq_lst(): Queue,Field
(test_queue,mohdir): Unable to find MOH files (/var/kamailio/MOH//test.8.*)!
Or if files exist at that location check the permissions.
Regards,
Hi Ryan,
Where are your trunks !?
if your provider can just send calls to your IP address then just do IP
based authentication in Kamailio and once provider is authenticated relay
the call to the Internal PBX.
so with reference to the code here:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.
Hey Max,
You should know that pstn.gw_ip is not a keyword in Kamailio and it
could've been pstn.lol_gw or anything. What really matters is that you know
where this gw_ip is being used in configuration.
Somewhere in the kamailio.cfg you'll see a PSTN route which will be doing
some RURI checks and t
.
Maybe an easier function exists for this somewhere or an efficient approach
exists. Thats just how I did it.
Regards,
Sam
On Oct 21, 2015 10:30 AM, "Safdar Khan" wrote:
> Hi Sam,
>
> Will u please elaborate.Means from where to start.
>
> Safdar.
>
> On Wed, Oct 21,
Hi Safdar,
The way I did was save the $mb variable in db when the Method is MESSAGE.
Regards,
Sam
On Oct 21, 2015 8:09 AM, "Safdar Khan" wrote:
> How can i store messages in silo table not only offline also online.Demo
> code will be appreciated.waiting for response.
>
> ___
The kamailio.bindport is not used for the purpose here.
To change the listen port modify the parameter "Listen" or "port"
On Oct 3, 2015 3:01 AM, "amjad ali" wrote:
> Hi,
>
> I am scratching my head for one month to get this sorted and I have spent
> so many hours but couldn't figure out. Please
Hi,
The way youve described it seems like you are not routing anything at all
to the asterisk. It all depends on your comfiguration on how you handled
the call. Somehow Ive a feeling that asterisk is used only for voicemail
and is called only once the B party is not found in lookup(location)
functi
81 _dlg_ctx.iuid.h_id = dlg->h_id;
882 set_current_dialog(req, dlg);
883 dlg_release(dlg);
884
885 return 0;
886
887 error:
(gdb)
On Tue, Sep 8, 2015 at 4:44 PM, SamyGo wrote:
> Thanks Daniel for pointing out, so if I upgrade my Kamailio it should
at you have to do in kamailio.cfg or database.
>
> Meanwhile, we can see the relevant pieces of code from your specific
> version with following commands in gdb:
>
> frame 0
> list
>
> frame 4
> list
>
> Cheers,
> Daniel
>
>
> On 08/09/15 21:12, SamyGo
e where it actually
> crashed?
>
> Cheers,
> Daniel
>
>
> On 08/09/15 20:20, SamyGo wrote:
>
> Hi,
>
> I'm randomly getting crash in my Kamailio with an error in log files like
> this:
>
> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 19
> ALERT:
Hi,
I'm randomly getting crash in my Kamailio with an error in log files like
this:
[pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 19
ALERT: [main.c:775]: handle_sigs(): child process 6853 exited by a
signal 11
ALERT: [main.c:778]: handle_sigs(): core was generated
INFO: [main.c:790
Hi Bruce,
Get some ideas from here:
http://www.opentelecom.it/cluecon/ClueCon_2015_Load_Balancing_HA.pdf
Also using DNS SRV is a good idea for use afront of multiple active proxies.
BR,
Sammy
On Aug 27, 2015 1:53 PM, "Bruce Lefko" wrote:
> If I want to have multiple kamailio proxies in front of
Hi Jean,
Can you further explain your question, the steps you mentioend are already
handled by kamailio's module rtpproxy. Now, you mentioned change the SDP in
kamailio before rtpproxy function is calledI wonder how the device
would feel about it.
For example UA sent and INVITE/SDP saying I'm l
[MANAGE_REPLY] {
>
> if (has_body("application/sdp")) {
>
>rtpproxy_answer("ie");
>
> }
>
> }
>
>
>
> route[WITHINDLG] {
>
> if (!has_totag()) return;
>
> if (loose_route()) {
>
>
Hi,
Try starting rtpprpxy with a / in between the two IP addresses.
For example -l 1.1.1.1/2.2.2.2
Besides that it depends where you are placing your rtpproxy function.
BR,
Sammy
On Aug 13, 2015 8:36 AM, "Grant Bagdasarian" wrote:
> Hello,
>
>
>
> I’m using RTPproxy for the first time in bridged
issue is , in the pcap file it is coming SIP/SDP as protocol and
> it is not coming SIP-I.
>
> Please find the latest attached pcap.
>
> Regards,
> Sandeep
>
>
> Warm Regards,
> Sandeep Chakravarthi.
>
> On Tue, Aug 11, 2015 at 9:47 PM, SamyGo wrote:
>
rd the SIP invite packet
> to MSC which it got from Asterisk server. But it is not happening.
> I am attaching the pcap one more time for your reference.
>
> In my pcap, below are the server details
>
> 172.22.14.12 - Kamailio server
> 172.22.14.17 - Asterisk server
> 172.22.0
8(1186) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg:
> msg id=2 global id=2 T start=0xb5d3f20c
> 8(1186) DEBUG: tm [t_lookup.c:1144]: t_check_msg(): DEBUG: t_check_msg: T
> already found!
> 8(1186) DEBUG: [msg_translator.c:205]: check_via_address():
> check_via_
Hi,
I'm working with IMC module and noticed that when I execute fifo command
"imc_list_rooms" it crashes.
[root@kamailio75 ]# kamctl fifo imc_list_rooms
my Kamailio version is 4.2.1
Right now the way IMC module works is that a SIP UA has to send a chat room
command to create a new room, get part
Thanks Daniel for pointing out.
On Mon, Aug 3, 2015 at 6:30 AM, Daniel-Constantin Mierla
wrote:
> Hello,
>
> On 01/08/15 16:03, SamyGo wrote:
> > Hi,
> >
> > Tried finding anything as weird as I am trying to do on mailing list
> > but couldn't. The ide
Hi,
Tried finding anything as weird as I am trying to do on mailing list but
couldn't. The idea is to save a copy of the messages on database.
MSILO module can keep the message as long as the destination user is
offline, or doesn't support method:MESSAGE or expiry timeout.
Any ideas will be appr
il.com> wrote:
> Hi,
>
> Can you share the sample code to differentiate the both telco IP and our
> server IP?
>
> .
>
>
>
> Warm Regards,
> Sandeep Chakravarthi.
>
> On Tue, Jul 14, 2015 at 10:55 PM, SamyGo wrote:
>
>> Sure but if you look into the d
th it.
>
>
>
>
> Warm Regards,
> Sandeep Chakravarthi.
>
> On Tue, Jul 14, 2015 at 10:28 PM, SamyGo wrote:
>
>> Hi,
>> You're right about using IP Auth in Kamailio. You'll need to use the
>> permissions module. However I believe permissions mo
Hi,
You're right about using IP Auth in Kamailio. You'll need to use the
permissions module. However I believe permissions module wont send the
OPTIONS to the MSC SIP Server. For this you may alternatively use the
"dispatcher" module.
Take a look at the sample kamailio.cfg here:
http://kb.asipto.c
I believe thats the configuration you've done via 'crm' to monitor for 3
crashes (by default probably) - that means if you kill it it will try start
the monitored application on the same node first and if fails in 3 attempts
only then goes to the secondary site. (try inducing an error in
kamailio.c
in subsequent BYE.
QUESTION: Was this supposed to happen with mhomed Kamailio ? I have not
observed this scenario with any provider/gateway other than this CISCO gw!
BR,
Sammy.
On Tue, May 12, 2015 at 12:24 PM, SamyGo wrote:
> Yeah sure, attached here is the trace from the CISCO logs. Not
://www.csrpswitch.com/
>
> Sent from my BlackBerry.
> *From: *SamyGo
> *Sent: *Tuesday, May 12, 2015 11:46
> *To: *SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List
> *Reply To: *Kamailio (SER) - Users Mailing List
> *Subject: *[SR-Users] Multi-ho
Hi all,
I'm having a scenario where I'm sending call to CISCO IAD2431. The call
establishes with 200OK from CISCO and Kamailio sends back modified ACK.
Cisco at this points gives out an error and sends 200OK again, Kamailio
replies with ACK, and this cycle goes on until the call drops.
Here is th
They should be unrelated, RTPPROXY called on INVITES and MSILO on
REGISTER|MESSAGE, why should INVITE's flow change using that msilo route ?
On Thu, Apr 30, 2015 at 4:13 AM, sscc wrote:
> # - msilo params -
> #!ifdef WITH_MSILO
>
> modparam("msilo", "db_url", "mysql://kamailio:abc@local
. When I
> log the header again after the changes it still shows me the old values.
>
>
>
> *From:* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
> Behalf Of *SamyGo
> *Sent:* Thursday, April 30, 2015 6:50 PM
>
> *To:* Kamailio (SER) - Users Mailing Li
voipfone.net";
>
> t_relay();
>
>
>
> }
>
>
>
>
>
> failure_route[F_VOIP] {
>
> uac_auth();
>
> xlog("L_INFO","ALERT: IN FAIL");
>
>}
>
>
>
>
>
> I tried this but it never ma
Could you provide your section of the code where you're using msilo ? how
you're using it !
I just wonder how could it just tell kamailio to skip rtpproxy stuff !!
On Wed, Apr 29, 2015 at 11:55 PM, sscc wrote:
> i have configured msilo module successful but there isn't any voice with
> msilo. i
Hi,
The important thing to consider here is this line.
#!define WITH_ASTERISK
so if you've defined this on the very top of your kamailio.cfg then it
will go and check username/passwords from the sipusers table from the
Database defined by this: DBASTURL
if (!auth_check("$fd", "sipusers", "1"))
Hi Jibran,
Here is an old thread as reference:
http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with
username/password on a Provider for huge number of calls..imagine sending
thousands of call to that pro
>From the diaries of the amazing Alex-Man,
Chapter 4 Page 1, Vol 15.
Congrats ;)
On Wed, Apr 1, 2015 at 12:11 AM, Alex Balashov
wrote:
> For immediate release:
>
> ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software
> vendor specialising in Kamailio-based service deliver
en reply
> }
>
> Similar you can add in request_route for sip traffic to be on port 5060
> only.
>
> Cheers,
> Daniel
>
>
> On 12/03/15 19:58, SamyGo wrote:
>
> Hi All,
>
> I am just trying xhttp_rpc module
> http://www.kamailio.org/docs/modules/4.2.x/module
Hi All,
I am just trying xhttp_rpc module
http://www.kamailio.org/docs/modules/4.2.x/modules/xhttp_rpc.html
Where can I set/change the port for this module ? currently its only 5060 !?
Thanks,
Sammy.
___
SIP Express Router (SER) and Kamailio (OpenSER)
Hi All,
Im trying to load up module 'mi_xmlrpc.so' into my v4.2.1 Kamailio. It
compiles nicely but when I try to load the module in config and restart
kamailio it gives following error(s)
0(30917) ERROR: [sr_module.c:597]: load_module(): could not open
module :
/usr/local/lib64/kamailio/modules/
>
> ok, good that is working better now.
>
> I will also look more into these module in the near future. I think the
> Expires value must be different based on state, e.g., trying and ringing
> should be shorter than active call.
>
> Cheers,
> Daniel
>
>
> On 1
t; readme for hints on its value.
>
> Also, force_single_dialog set to 1 could be considered, but lowering the
> lifetime should make it work.
>
> Let me know if works ok with these settings.
>
> Cheers,
> Daniel
>
>
> On 07/02/15 10:48, SamyGo wrote:
>
> Please
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