sr-users January 2016

sr-users@lists.kamailio.org

86 participants
111 discussions

A new edition of Kamailio Development Workshop
by Daniel-Constantin Mierla 27 Jan '16

27 Jan '16

Hello, last edition of Kamailio Development Workshop happened about three years ago. As I got requests about it from time to time, I thought of trying to organize a new edition. Rather short term by now due to various constraints, so the plan is to do it again in Alicante, Spain, during February 15-16, 2016. I put together more details at: * http://www.kamailio.org/w/development-workshop/ Besides spending 2 days digging into Kamailio C code, such event is also good for discovering and … [View More]

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Acc message to syslog
by Yasin CANER 27 Jan '16

27 Jan '16

4 3

Strange PUA Behaviour
by Phil Lavin 26 Jan '16

26 Jan '16

Hi all, We are investigating an issue with BLF whereby the BLF key on the subscribed phone does not always "turn off" after the call has ended. We have enabled force_single_dialog to keep things simple for debugging. The attached pcap is the result. The relevant stuff as I see it is: * Packet #60 - UA sends a BYE * Packet #62 - PUA sends a PUBLISH over loopback with a state of "terminated" * Packet #65 - The PUBLISH is sent out to the subscribed UA as a NOTIFY with … [View More]

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IPTel URI forwarding to PBXes not working
by McConnachie, Alex 26 Jan '16

26 Jan '16

I am stumped as my DID provider can't route via SIP URI to PBXes but they can to IPTel. I can call my own PBX, and from my own PBX to the PBXes URI without issue. When I route the DID via URI to IPTel and set voicebox it works and routes to the voicebox. When I add a forward for the PBXes URI it then breaks and gets a denial of 404 not in service. Ideas anyone? I am scratching my head here. Appreciate any help! Regards, Alex McConnachie

2 1

Rewriting header of BYE message to carrier
by Ryan Mottley 26 Jan '16

26 Jan '16

Hi, We're running a system with Kamailio running in front of Asterisk just handling registrations and forwarding everything else to Asterisk. But we're having an issue during hangup on incoming calls. If the initiator hangs up, the call completes successfully. But if one of our phones hangs up, the BYE message comes back with a 404 "Not Found" and the call doesn't hang up on the carrier side. According to the carrier, it's because the IP in the contact on our ACK message goes to their audio … [View More]

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Missing dependencies for kamailio-mysql-4.2.7-3.1.x86_64
by Zuxy Meng 26 Jan '16

26 Jan '16

Hi there, I'm trying to install this rpm for my RHEL 5 box, but it asks for mysql-libs; but RHEL 5 isn't shipped with mysql-lib, only mysql55-mysql-libs, which kamailio-mysql doesn't recognize as a valid dependency. Any workaround here? Thanks! -- Zuxy Beauty is truth, While truth is beauty. PGP KeyID: E8555ED6

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Fwd: [Free-RTC] FOSDEM Real Time devoom: call for help
by Daniel Pocock 26 Jan '16

26 Jan '16

-------- Forwarded Message -------- Subject: [Free-RTC] FOSDEM Real Time devoom: call for help Date: Sun, 24 Jan 2016 19:17:02 +0100 From: Saúl Ibarra Corretgé <saul(a)ag-projects.com> Reply-To: Free real-time-communications discussion list <free-rtc(a)lists.fsfe.org> To: Free real-time-communications discussion list <free-rtc(a)lists.fsfe.org>, summit(a)xmpp.org Hi all, FOSDEM is less than a week away, I hope you are all as excited about it as I am :-) We need some … [View More]

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question about kamailio handling of forwarded invite
by DING MA 26 Jan '16

26 Jan '16

Hi, all We're trying to build a system that consists of pbx, kamailio and asterisk in the following configuration. pbx (sip trunk) --- kamailio --- asterisk The kamailio and asterisk are integrated with same database. The outgoing calls to pbx works. But there is a problem with incoming calls from pbx. If we make a consecutive calls from the same pbx user to the same user registered with kamailio. The first would go through, but the second call would be rejected by asterisk. We have insecure=… [View More]

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Acc leaking
by Yasin CANER 25 Jan '16

25 Jan '16

3 3

kamailio_call_disconnecting issue
by yasir ay 25 Jan '16

25 Jan '16

Hello, Hi team I have a kamailio SIP server for call Routing,my call flow is like VoipDialer--->Kamailio-Routing Server-->Softswitch-->Telco/PSTN Mobile caller----------------------------------------------------------------------->calle Now Am facing an issue for a call from voip to telco after answering the call i tried to disconnect.but call is not disconneting..what will i do in kamailio.cfg pls help me to re-write the script am using tls also Regards AmanIdo

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sr-users January 2016