sr-users June 2016

sr-users@lists.kamailio.org

85 participants
126 discussions

uac.reg_refresh not removing records from memory
by DanB 16 Jun '16

16 Jun '16

Hey Guys, I was wondering if there is any dynamic way today to remove a single registration out of memory (and implicitly to force un-REGISTER) Eg: I add one record using uac.reg_refresh (which works fine) but after removing the record from the database and executing uac.reg_refresh again, the record stays in memory until expire. Is there a better way to enforce un-register on remote side? Thanks in advance! DanB

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Question about testing RTPENGINE
by Dmitry 16 Jun '16

16 Jun '16

Hello I am trying to make a test setup to test the RTPENGINE. Case 1: Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.136)-----via NAT(10.111.101.3)-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) Case 2: When A calls to B - the sound both ways, so it is ok. Subsriber A ---Asterisk (10.20.20.101)-----via NAT(10.11.97.12)-------------Kamailio(10.10.11.104) ----… [View More]

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RTPENGINE not kernalizing in some cases
by Dmitry 16 Jun '16

16 Jun '16

Hi I have the Kamailio and FS behind it (in Kazoo). Both are in one ip address 10.34.101.104 i use rtpengine_manage(). When I call from Asterisk which is behind NAT :1) Asterisk in VMware player (NAT, local ip 192.168.175.136) and public 10.34.101.103 - the voice was ok? in logs I saw Kernalizing media 2)Asterisk in KVM(NAT, local 10.28.40.101 public 10.101.29.45) and no Kernalizing in /var/log/messages and no voice. The Asterisk configs are absolutely the same. Kamailio log … [View More]

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HTTP query at the end of a phone call
by Jay Li 15 Jun '16

15 Jun '16

Dear All, I'd like to send a HTTP call to a HTTP server whenever a call routed through a Kamailio server ends. By looking at the document, seems I should use http_query() function in the utils module. I wonder which route function I should put the http_query() in assuming we use the default kamailio.cfg. Thanks. Regards,Jay

4 3

404", "Not Found location"
by Sébastien Brice 15 Jun '16

15 Jun '16

Hi guys when i try to register i have route[LOCATION] that returns "404 Not Found location" due to this statement: $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found location"); exit; … [View More]

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Kamailio + ToS
by Marcelo Bonin 15 Jun '16

15 Jun '16

Hello, I need to know what are the available parameters to set ToS on Kamailio. I couldn't found much information about that, only this https://www.kamailio.org/wiki/cookbooks/4.0.x/core#tos but there is no reference for the available parameters. Thanks, Marcelo.

2 1

question about RTPengine
by Dmitry 14 Jun '16

14 Jun '16

hello i read the documentation about RTPengine.and the documentation says that: If INVITE with SDP, when the tm module is loaded, mark transaction with internal flag FL_SDP_BODY to know that the 1xx and 2xx are for rtpengine_answer() what does it mean? actually - according to 3261 RTP media (both ways) may be turned on after ACK

3 5

IP TOS
by Daniel W. Graham 14 Jun '16

14 Jun '16

Could I get some clarification on tos global setting as referenced by documentation below? Specifically, are DSCP values allowed? Or is this simply for scheduling purposes at the OS level? I can change the value once packets hit the network but if possible would like 0x60 to originate from Kamailio. tos The TOS (Type Of Service) to be used for the sent IP packages (both TCP and UDP). Example of usage: tos=IPTOS_LOWDELAY tos=0x10 tos=IPTOS_RELIABILITY Thanks! -Dan

2 3

Fwd: [Free-RTC] GSoC progress, more mentors?
by Daniel Pocock 14 Jun '16

14 Jun '16

This report was sent to the Free-RTC list https://lists.fsfe.org/mailman/listinfo/free-rtc Most of the projects have an emphasis on SIP If anybody wants to facilitate testing the new Telepathy/Empathy SIP connection manager code against Kamailio or other SIP proxies, registrars and presence servers that would be very welcome. -------- Forwarded Message -------- Subject: [Free-RTC] GSoC progress, more mentors? Date: Mon, 13 Jun 2016 22:53:46 +0200 From: Daniel Pocock <daniel(a)pocock.pro… [View More]

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Kamilio as Scalable Registrar + Proxy
by Colin Morelli 13 Jun '16

13 Jun '16

Hello, I currently have Freeswitch acting as a B2BUA, handling registrations, routing, etc for a prototype voice application I have built. Now I'm at the point where it's time to actually scale it out, and I'm looking at a few different options for the SIP proxy + registration later. Essentially, what I would like to have (I think), is: 1) Stateless SIP proxy with consistent hashing to route to: 2) Transaction-stateful SIP proxy handling registrations and initial routing Ideally, the only … [View More]

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sr-users June 2016