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37517 discussions

Scaling websockets over SIP
by Jade SZ 01 Mar '17

01 Mar '17

Hi Guys, I am using kamailio/rtpengine and webrtc over sip client for a/v calls and all works fine. Now I want to scale it to 2 or more than 2 kamailio servers. As per search on mailing list I could: 1) Websocket Outbound/Edge Proxy 2) Each UA client to register with its Regards, Jade SZ

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pushing alarms/counters other than with SNMPstats
by Valentin Christoph 28 Feb '17

28 Feb '17

Hi all I'm looking for ways to push stats/alarms from kamailio to some external destination other than using the SNMPstats module. Any ideas? Thanks Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

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sip over websockets sip uri
by Jade SZ 28 Feb '17

28 Feb '17

Hi Team, Question: Just wanted to clarify regarding SIP URI in webrtc (over sip) connection. e.g. if we have a scenario where Kamailio is hosted with websocket support. Websocket URI is used to send packets to to wss/ws address and the SIP URI goes with it. I have tested it with any SIP URI including any domain and it works if I authenticate that domain, regardless that domain is real or not. // client side SIP URI config var configuration = { uri : 'sip:alice@example.com', }; Can we use any central domain for sip uri similar to realm concept and send traffic to multiple kamailio servers through websocket LB. dummy.domain - for ws/wss SIP URI kamailio-a.domain.com kamailio-b.domain.com Thanks in advance. Regards, Jade

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Newbie starting - looking for docs / tutorials
by Antony Stone 28 Feb '17

28 Feb '17

Hi. I'm just starting with Kamailio - I've been to a few presentations about it, at Fosdem and KamailioWorld, but never used it so far. I'm pretty familiar with Asterisk, and I want to use them in combination. My first use case is to handle SIP registrations (both as a client, registering to a remote SIP server, and as a server, accepting registrations from handsets) in Kamailio instead of Asterisk, and still be able to route calls to/from the registered accounts through the Asterisk dialplan. I suspect this is a pretty common starting point for people who've used Asterisk and want to add Kamailio to the setup? So far I've been unsuccessful finding any Howto or Tutorial explaining how to do this - I can't help feeling I must simply be looking in the wrong place or in the wrong way. Searching for topics like "Asterisk Kamailio Getting Started" I find: http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf but with 12 chapters and 85 pages I'm not sure which bits are the essential parts to do what I need. I've also found http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x- asterisk-11.3.0-astdb but this seems to focus on a database backend, which I guess I'll want to use in due course, but doesn't seem the simplest place to start just to learn how Asterisk & Kamailio work together? So, please can someone suggest a beginner's guide to introducing Kamailio into an Asterisk setup, keeping things as simple as possible to start with, so I can learn about Kamailio one bit at a time (since there do seem to be a lot of bits to it) from a familiar environment of using it with Asterisk? Thanks in advance, Antony. -- What do you get when you cross a joke with a rhetorical question? Please reply to the list; please *don't* CC me.

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multi domain kamailio
by przeqpiciel 28 Feb '17

28 Feb '17

Let's suppose that i have two machines with installed asterisk and one with kamailio. I would like have routing based on sip domain. for domain sip.domain1.com send sip signalling to asterisk#1 server and for sip.domain2.com send to asterisk#2 server. What i should use? Domain module or what ? Tomasz

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Kamailio v5.0.0 Released - new major version is out
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Kamailio v5.0.0 is out – it comes with 6 new modules and a consistent set of improvements touching more than 50 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-0-0-release-notes/ Many thanks to all developers and community members that made possible this release. A consistent effort was directed to restructuring the source code tree to meet more modern patterns; introducing the KEMI framework that allows writing SIP routing blocks in other embedded interpreters such as Lua, JavaScript or Python; removal of obsoleted MI control mechanism, being replaced by RPC. Enjoy Kamailio v5.0.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference 2017 - http://www.kamailioworld.com

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problem with avpops and mongodb modules
by hdssdsdsdsfsdf hdssdsdsdsfsdf 27 Feb '17

27 Feb '17

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Releasing Kamailio v5.0.0
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Hello, I am considering to release Kamailio v5.0.0 at the beginning of next week, depending how things evolve in the next days, either on Monday, Feb 27, or Tuesday, Feb 28. I haven't noticed any new issues specific for the new features added for 5.0 lately, so it looks pretty good so far. If you are aware of issues not reported to the bug tracker, register them as soon as possible at: - https://github.com/kamailio/kamailio/issues There are still some parts of documentation that have to be updated, I am planning to work on them during the next days, but help from others are very appreciated, especially details of what you had to do changes while upgrading from 4.4 to 5.0. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Parallel Forking with Different Call-Id
by SamyGo 24 Feb '17

24 Feb '17

Hi, I've a very strange scenario to work on which requires me to parallel fork the call to the same Destination provider. The only problem here is that they think that the second INVITE with different branch tag is a re-transmission and hence only take one call forward. I do not have to modify any R-URI or any headers, hence just 'append_branch() before t_relay() is in the code. ... $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); append_branch(); route(RELAY); ... *Question:* Is there anyway possible I can change the CallID of the forked INVITEs ? I have tried using Topoh module, but it still puts the same CallID before sending out. Different from the A-leg but the Sent out INVITEs have same Call-ID value. In other weird scenario, I've also tried branching, and looping call within Kamailio before sending out to carrier hoping that Kamailio would treat the two different calls and Topoh would change the Call-ID on both INVITEs before sending out.. ... if(is_present_hf("X-FORKED")) { $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); route(RELAY); }else { append_hf("X-FORKED: 1\r\n"); append_branch(); route(RELAY); } ... Still I get same Call-ID on outgoing branched call. I could branch out one INVITE to a MediaServer, say FreeSWITCH/Asterisk but again I don't want to have that component bottleneck the throughput. That could be my very last option. Looking for some ideas. Regards, Sammy

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Kamailio 4.3.x location_attrs update issue
by Mridul B 24 Feb '17

24 Feb '17

Hi, We've noticed a case where the *location_attrs* table isn't updated when there's an unexpected shutdown of a SIP device (here there'll be no REGISTER message sent with expires=0). We're using Kamailio v4.3.x When the SIP device shuts down ungracefully. The location table is updated using DELETE query This is done by the timer which deletes the expired contacts, by default the timer_interval is set as 60. Here, the logs of auto-removal of expired contacts entries by the timer > Feb 17 17:16:13 localhost kamailio[23273]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:14' AND expires<>'1970-01-01 00:00:00'] > Feb 17 17:16:14 localhost kamailio[23272]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:15' AND expires<>'1970-01-01 00:00:00'] The delete operation is only on location table, shouldn't we have the same implementation on the location_attrs table based upon ruid or username? We're getting duplicate entries in the location_attrs table of the same sip user after it registers. But when the SIP device un-registers, the entries in the location as well as in the location_attrs table is deleted Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: usrloc [ucontact.c:1725]: > uldb_delete_attrs(): trying to delete location attributes > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location_attrs where username='mridul' > AND ruid='uloc-58a72243-5ace-2'] > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location where username='mridul' AND > contact='sip:mridul@127.0.0.1:37881;rinstance=b841a8c916e47dfa'] I think the usrloc module should have the implementation of deleting the entries of expired sip users based on its ruid/username in the location_attrs table also. Thanks for the help. Best regards, MRIDUL

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