sr-users August 2015

sr-users@lists.kamailio.org

79 participants
123 discussions

Back to the dead communications: No outgoing audio

by Manuel Camargo Lominchar

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 checks as before. I started to assume that rtp-engine could be failing by the same cause, but I have updated to the latest version and still the same issue This has happened from 2-3 day to now. Any ideas on how to start trying to find the cause? Versions: Kamailio 4.1.4 RTP Engine: 4.1.0.0 Kind regards, *Manuel Camargo* Teléfono: 638000836 eMail: sir.louen(a)gmail.com

9 years, 1 month

Mitel 5000 and 3000

by Fred Posner

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)

9 years, 1 month

Re: [SR-Users] Mitel 5000 and 3000

by rahul.ultimate

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 with the media server vs the sip server; as well as general audio issues in/out. Most of these seem to be acl related with a set-up issue on the Mitel. I don't work with Mitel enough to discuss in "Mitel speak" what to change. Apparently RTP ports is a different language to the people I am working with. =) --fred _______________________________________________ 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

9 years, 1 month

new SIP account can not be created ... DB Error : connect failed

by Rainer Böing

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 THANKS Greetings from Germany Rainer Böing Disclaimer: Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

9 years, 1 month

Change ports for outgoing RTP packets in RTPproxy

by Jean-Marie Baran

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*

9 years, 1 month

how to use seturi with varibales in kamailio

by SureshTummala

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

9 years, 2 months

Kamailio with NATted RTPproxy

by Jean-Marie Baran

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 the IP in o= and c= by its advertising IP (i.e. its public IP), so the packets from the client behind the NAT were sent to the public address of Kamailio (and then dropped). My question here is what should be the addresses in the SDP when redirected from Kamailio to the client ? Here is the topology when receiving an ACK with SDP: Client <--(2)-- Kamailio <--(1)-- Router (NAT) <--(1)-- Public Internet <--(1)-- Third party router (21.24.12.24) (1): SDP o= 21.24.12.24, c= 21.24.12.24 (2): SDP o= ??, c= ?? (Should it be Kamailio public IP (actually Router IP), or Kamailio private IP, or third party SIP server public IP ?) Cheer, -- *Jean-Marie Baran*

9 years, 2 months

rtpengine rewrite sdp for second time

by Virmantas Variakojis

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] 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 > >

9 years, 2 months

Routing re-INVITEs after phone port change

by David Cunningham

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 Kamailio should send the re-INVITE to the new address/port, but is not. The re-INVITE is routed using the lookup() function. Can anyone suggest why this is happening? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

9 years, 2 months

Kamailio/Asterisk Integration - MySQL driver error

by Ben Fitzgerald

Has anyone seen this error before? " ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Host '192.168.xxx.xxx is blocked because of many connection errors; unblock with 'mysqladmin flush-hosts'" I fixed it using flush hosts but I'm curious why this popped up? Is there a way to limit connections for Kamailio and why am I reaching that limit with *only 2* SIP endpoints? This is not a heavily utilized database by any means. Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us *******Confidential Notice: This message is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this message in error, please delete this message from all computers and contact Orion Systems/LETS Corp immediately by return e-mail and/or telephone at (925) 566-5600

9 years, 2 months

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sr-users August 2015