sr-users August 2015

sr-users@lists.kamailio.org

79 participants
124 discussions

Presence and NAT
by Pramod Venugopal 21 Aug '15

21 Aug '15

Hello everyone, I have a Kamailio 4.2 system on Amazon AWS. When subscribing to presence the contact is sent out with the internal IP rather than the external public IP. The subscribes are being sent to it with the external hostname ( which resolves to the external IP ). Where would I be able to change the IP sent in the Contact header ? Pramod Venugopal pramod(a)perseus-tech.com

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Back to the dead communications: No outgoing audio
by Manuel Camargo Lominchar 20 Aug '15

20 Aug '15

After 6 months of resolving this situation: http://lists.sip-router.org/pipermail/sr-users/2015-February/086980.html Without touching any configs, my server is back to a failure situation I'm not sure if I'm facing the exact same thing, but for some strange reason, without introducing any modification in the servers, one leg is not hearing voice (outgoing traffic), while incoming rtp traffic goes fine, so I have to assume another time that some Update has destroyed it I've mostly done the same … [View More]

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Mitel 5000 and 3000
by Fred Posner 20 Aug '15

20 Aug '15

I rarely integrate these with Kamailio and am having some "resistance" in assisting. I was wondering if anyone would be willing to share any configuration recommendations for Mitel when using Kamailio as a SIP trunk (outbound/inbound). Sincerely, Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

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Re: [SR-Users] Mitel 5000 and 3000
by rahul.ultimate 20 Aug '15

20 Aug '15

Hi Fred I too am new to Mitel products but the attached pic might be of some use to you. Thanks RM Sent from Samsung Mobile -------- Original message -------- From: Fred Posner <fred(a)palner.com> Date:18/08/2015 7:08 PM (GMT+05:30) To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Mitel 5000 and 3000 On 08/18/2015 09:23 AM, Alex Balashov wrote: > Are there any specific issues you're encountering? > Mostly hold issues where they try to speak directly … [View More]

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new SIP account can not be created ... DB Error : connect failed
by Rainer Böing 20 Aug '15

20 Aug '15

Hello, want to build the following account about Ipel.Org . My old following account no longer works. Iptel.org 19.11.2013 E-Mail rainer.boeing(a)t-online.de Username: boe.rai(a)iptel.org <mailto:boe.rai@iptel.org> Password: xd7ghg Sip address: <sip:boe.rai@iptel.org> sip:boe.rai@iptel.org I can also create no new no account . Please Help !!! clamp … [View More]

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Change ports for outgoing RTP packets in RTPproxy
by Jean-Marie Baran 19 Aug '15

19 Aug '15

Hi, Is there a way to change the ports were RTPproxy sends outgoing packets ? By rewriting the SDP before it gets to RTPproxy, or anything else ? 1 - Kamailio receives the SDP 2 - It somehow transmits the contact information to RTPproxy (ip + ports) 3 - The client start sending packets to RTPproxy which in turn relays them to the contact How to change the ports in step 2 ? Sincerely, -- *Jean-Marie Baran*

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how to use seturi with varibales in kamailio
by SureshTummala 19 Aug '15

19 Aug '15

Hi, I have the following questions 1. How to remove ruri which is already added to the destination set2. How to rewritehost in ruri with variables3. How to change the To header in INVITE which are generated by t_relay based on destination set Regards,Suresh Tummala

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Kamailio with NATted RTPproxy
by Jean-Marie Baran 18 Aug '15

18 Aug '15

Hi, I am trying to have RTPproxy works with Kamailio behind a NAT. As it seems RTPproxy cannot handle NAT natively, I installed RTPproxy 1.2.1 by miconda: https://github.com/miconda/rtpproxy Now I am able to make calls, but neither the video nor the audio are present. Wireshark showed that the RTP packets were sent to wrong IPs (private IPs). And this was because of the SDP which contained the wrong IP (fields o= and c=). Then I tried to play with rtpproxy_manage(), and Kamailio replaced … [View More]

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rtpengine rewrite sdp for second time
by Virmantas Variakojis 18 Aug '15

18 Aug '15

Hi Carsten, Thanks for your reply! How can I call msg_apply_changes() in right way? If I call msg_apply_changes() for example in route[RELAY] i get error: ERROR: textopsx [textopsx.c:157]: msg_apply_changes_f(): invalid usage - not in request route if I do everything as you suggested in onreply_route (after I get for example SIP 500 message) it doesn't work and my SDP remains unchanged. Thanks! > > *From: *Carsten Bock <carsten(a)ng-voice.com> > *Subject: **Re: [SR-Users] … [View More]rtpengine rewrite sdp for second time* > *Date: *17 Aug 2015 09:49:12 EEST > *To: *"Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org > > > *Reply-To: *"Kamailio \(SER\) - Users Mailing List" < > sr-users(a)lists.sip-router.org> > > Hi Virmantas, > > you could try to one or a combination of the following: > - explicitly delete the first session by calling rtpengine_manage() in > the failure route > - call msg_apply_changes() from Textops before calling > rtpengine_manage() the second time (so the already made changes are > applied and the message gets reparsed) > > Thanks, > Carsten > > 2015-08-16 20:13 GMT+02:00 Virmantas Variakojis > <virmantas.variakojis(a)gmail.com>: > > Hi, > > I have a situation like this: > I have kamailio with 3 NIC's. I'm sending INVITE to first provider over > NIC1. Before sending INVITE I use rtpengine_manage to rewrite sdp body with > IP that is on NIC1: > rtpengine_manage ("replace-origin replace-session-connection > direction=intvlan direction=prov1 ICE=remove"); > > If first provider fails i want to send INVITE for the same call to second > provider. But if i use like this: > rtpengine_manage ("replace-origin replace-session-connection > direction=intvlan direction=prov2 ICE=remove"); > or (because ip of prov1 is allready set): > rtpengine_manage ("replace-origin replace-session-connection > direction=prov1 > direction=prov2 ICE=remove"); > rtpengine ignores my changes as said in documentation: "The direction must > only be specified in for initial SDP offer; answers or subsequent offers > can > omit this option." > > and if I try to set ip like this: > rtpengine_manage ("replace-origin replace-session-connection > media-address=X.X.X.X ICE=remove"); > SDP body is doubled: it leaves old values and adds the new one. > > What i'm doing wrong or how can I rewrite SDP for second time? Thanks in > advance for answer! > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > -- > Carsten Bock > CEO (Geschäftsführer) > > ng-voice GmbH > Schomburgstr. 80 > D-22767 Hamburg / Germany > > http://www.ng-voice.com > mailto:carsten@ng-voice.com > > Office +49 40 5247593-0 > Fax +49 40 5247593-99 > > Sitz der Gesellschaft: Hamburg > Registergericht: Amtsgericht Hamburg, HRB 120189 > Geschäftsführer: Carsten Bock > Ust-ID: DE279344284 > > Hier finden Sie unsere handelsrechtlichen Pflichtangaben: > http://www.ng-voice.com/imprint/ > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > [View Less]

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Routing re-INVITEs after phone port change
by David Cunningham 18 Aug '15

18 Aug '15

Hello, We have a scenario like this: SUA -> Kamailio with registrar module -> Asterisk A call from the SUA is set up with SIP timers, and after 15 minutes Asterisk sends a re-INVITE to Kamailio to forward on to the SUA. That re-INVITE has a RURI with the address and port of the SUA at the time the call started. Now if the SUA re-registers after the call starts and before the re-INVITE, and is on a new address or port number, then the re-INVITE never gets to the phone. Obviously … [View More]

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sr-users August 2015