sr-users May 2017

sr-users@lists.kamailio.org

91 participants
117 discussions

MSILO DUMPING SAME MESSAGE MULTIPLE TIMES
by sagar malam 15 May '17

15 May '17

Hello, I have been using Kamailio 4.4.In there i have used MSILO module to store messages when B party is not available and dump message when it gets registered. But when user gets registered,MSILO is dumping same messages multiple times. On further observation i found that MSILO stops dumping message when 200 OK for any one of the multiple message requests arrives to server.Hence i tried configuring "check_time" parameter to higher value but it did not help. I am am using same example script as provided in below link : http://kamailio.org/docs/modules/4.4.x/modules/msilo.html Except tm module parameters. Please help me on this issue. Thanks in advance.

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IP Address used for outgoing Packets?
by Keith Whyte 15 May '17

15 May '17

Hi All, tl;dr - Kamailio is sending traffic to the Voip server with a source IP of the private (vpn) interface. For some weeks, I've been experimenting with Kamailio v 4.4.4, trying different options, including running (or not) a vpn interface on the kamailio server with and without NAT/rtpproxy. I have not as yet fully "installed" kamailo with database etc, I have just been playing with the config file, mostly running with -D For most of the configurations I tried, I was routing SIP to/from the VPN via another box on the network, and I tried to get RTP streams running without using rtpproxy, sometimes it worked, but I have two many double/triple NAT problems with some of the freeswitch nodes. Now I have decided to run a VPN interface directly on the kamailio server, for simplicity. Essentially what I want to do is have kamailio/rtpproxy bridge between the VPN and the VOIP provider, so it should be a pretty standard NAT config. My setup looks like this: VPN CLUSTER of Freeswitch nodes <-10.23.0.1-> KAMAILIO Server <-192.168.1.1 - gateway - PUBLIC_IP-> Voip Provider The Kamailio Server has two interfaces, one VPN interface and one facing the Public Internet. It is behind a firewall, and so has a private IP address assigned on the interface, but the is dedicated public IP that forwards all relevant ports, and traffic seen on the public side originates from this address. I have two listen directives, listen=udp:192.168.1.1:5060 advertise PUBLIC_IP:5060 listen=udp:10.23.0.1:5060 I expected Kamailio to receive SIP INVITE on the 10.23.0.1 address and forward it to the voip provider via the default route (the 192.168.1.1 address), but it is sending the packet out to the voip provider with a source address of 10.23.0.1 !! Can anyone tell me how I can configure this? I cannot find anything obvious in the manual. Thanks!

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SIREMIS 4.3.0 and Kamilio 5.1.0-dev1
by Emil Morganti 12 May '17

12 May '17

Hi everybody, I am installing siremis 4.3.0 with kamilio 5.1.0-dev1 but when i step to the second step i get an error: ERROR: SQLSTATE[23000]: Integrity constraint violation: 1062 Duplicate entry '127.0.0.1' for key 'domain_idx' Some solution?

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TLS to TCP
by David Villasmil 12 May '17

12 May '17

Hello Guys, I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on tcp, but AFTER the call is established, kamailio hangs the call with "SIPS required"... Has this happened to anyone? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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t_drop_replies not working with t_suspend in failure route
by Huber Andreas 12 May '17

12 May '17

Hello, We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE. We'd like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect. The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch. But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something? Kind Regards, Andreas

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Re: [SR-Users] Can rtpproxy stream audio inside a call? (Maxim Sobolev)
by Yufei Tao 12 May '17

12 May '17

Hi, Thank you very much for the answers! I'll have a play. Cheers, Yufei Message: 9 Date: Mon, 8 May 2017 01:01:51 -0700 From: Maxim Sobolev <sobomax(a)sippysoft.com> To: sr-users(a)lists.kamailio.org, Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: Re: [SR-Users] Can rtpproxy stream audio inside a call? Message-ID: <CAH7qZfs0tQTwa-aCwvHHhAS_hajCRNaKBLnjXBv+O0txJNd4Mw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Daniel is right. Technically speaking it can be done at any point after the media session has been established, but you need some kind of trigger to start/stop it from your routing script. Such as re-INVITE for example. -Max On May 8, 2017 7:52 AM, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: > Hello, > > it can play rtp files encoded in the format expected by the audio codec. I > used only when there was a re-invite (like putting the call on hold). > > Cheers, > Daniel > > On 04.05.17 17:29, Yufei Tao wrote: > > Hi, > > Just a quick question: if I use the Kamailio rtpproxy module with the Sippy > RTPproxy, can I stream audio within a call? Or is it only possible before > call audio is established? > > For example can I mix/playback audio file into an existing established > call session at any time using functions rtpproxy_stream2uac() and rtpproxy_stream2uas() > somehow? If it is possible, does it need to be triggered by an in-dialog > message from either UAC or UAS? > > Cheers, > Yufei > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps:// lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda <http://mierlawww.twitter.com/miconda> -- www.linkedin.com/in/miconda > Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com > Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >

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Config Kamailio with SEMS
by Nguyen Tran Nhan 11 May '17

11 May '17

Dear Daniel, I am working on Push To Talk PoC. I intent to use Kamailio as SIP signaling to work with SEMS for handling mixer conference audio (media server). The reason to use SEMS is performance is better in comparing with FreeSwitch or Asterisk. There are some guide to config Kamailio with Freeswitch or Asterisk but no for SEMS. If you have any information or guide line, please help!

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Warning "constant value in if(...)"
by Ginhoux, Patrick 11 May '17

11 May '17

Hi, I discovered that when kamailio (version 5.0.1) starts, if reports a waring on a 'if' command : May 11 11:26:20 vm-vse02-siprouter2 kamailio: WARNING: <core> [core/cfg.y:3378]: warn_at(): warning in config file //etc/kamailio/kamailio.cfg, line 978, column 6-13: constant value in if(...) The line 978 (in yellow) is part of the following routing below in my Kamailio.cfg : route[TO_MMM] { $var(mbxDN)=$null; # directory number containing mailbox number $var(uDigit)=$null; # application digits $var(i:NodePath)=$null; $var(TechNumber)=$null; $avp(nodeInx)=$null; # strip prefix from user in URI request user field $var(TechNumber) = $(rU{re.subst,/^(\+33|\+262|\+377|\+590|\+594|\+596|0)//}); # strip suffix from user in URI request user field - suffix headed by anything other than digit or '+' $var(TechNumber) = $(var(TechNumber){re.subst,/[^\+0-9].*//}); if (VM_TARGET_TECHNICAL_NUMBER!=$null) { $avp(vm_prefix) = VM_TARGET_TECHNICAL_NUMBER; } else { $avp(vm_prefix) = $(var(TechNumber){s.substr,0,5}); The same code with kamailio 3.3 doesn't report this warning. Also the same code exists in another part of the routine but don't report a warning. I have no idea of the impact of such warning. Also I'm looking for help to understand the problem. Thanks in advance for the answers. Cordialement Patrick GINHOUX

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URGENT! stateless dispatcher BYE issue
by Satish Patel 11 May '17

11 May '17

We have 1 stateless dispatcher and 3 proxy behind that dispatcher, when Proxy send BYE message then dispatcher bouncing back that BYE to Proxy and they playing ping pong. I think i am missing something.. How BYE get routed? in dispatcher i am not using loose_route() function also should i need to use that but again we running dispatcher in stateless. Here i put SIP trace: https://pastebin.com/vp9JCdNP Dispatcher config, Very simple.. route{ remove_hf("Route"); if (method == "INVITE") { record_route(); } if ( !ds_select_dst("1", "0") ) { xlog("L_ERR", "Unable to route \n"); sl_send_reply("500","Unable to route"); break; } forward(uri:host, uri:port);

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Conference Factory URI
by Nguyen Tran Nhan 11 May '17

11 May '17

Hi, I want to create conference without interact with IVR, so, trying to use Conference Factory URI. https://tools.ietf.org/html/rfc4579#section-3.2 I am looking for documentation but it seems very poor documentation. Could you please help to give more document or opensource for Conference Factory URI. SEMS does not support Conference Factory URI I guess? Thanks, Nhan

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sr-users May 2017