sr-users December 2014

sr-users@lists.kamailio.org

84 participants
124 discussions

rtpengine offer does not replace o and c line addresses
by jh＠tutpro.com 15 Dec '14

15 Dec '14

while doing some webrtc tests, i noticed that for some reason rtpengine_offer replace-session-connection replace-origin flags don't have any effect on the ip address (0.0.0.0) at o and c lines of the sdp. the debug is below. what is it that i'm missing? -- juha Dec 15 08:26:29 rautu /usr/bin/sip-proxy[12597]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1) Dec 15 08:26:29 rautu rtpengine[10473]: Got valid command from 127.0.0.1:38653: offer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-34.0 24971 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:fb2938b0#015#012a=ice-pwd:eb43995297ab91228db0dec8949497ed#015#012a=fingerprint:sha-256 D9:DD:0F:CE:C2:D5:6A:D3:9A:C7:24:00:76:24:EA:66:ED:97:9F:E3:2D:25:23:60:67:7C:54:CF:04:8A:DB:F7#015#012m=audio 9 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8 ... Dec 15 08:26:29 rautu rtpengine[10473]: ... 000#015#012a=fmtp:101 0-15#015#012a=sendonly#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=setup:actpass#015#012a=rtcp-mux#015#012a=candidate:0 1 UDP 2122187007 192.98.102.30 58681 typ host#015#012a=candidate:2 1 UDP 2122252543 192.168.1.194 42724 typ host#015#012a=candidate:0 2 UDP 2122187006 192.98.102.30 53515 typ host#015#012a=candidate:2 2 UDP 2122252542 192.168.1.194 34867 typ host#015#012", "ICE": "force", "replace": [ "session-connection", "origin" ], "call-id": "dgp00b8p8q45f0vba38o", "via-branch": "z9hG4bK8492226", "rec ... Dec 15 08:26:29 rautu rtpengine[10473]: ... eived-from": [ "IP4", "192.98.102.30" ], "from-tag": "a4dr7vb4on", "command": "offer" } Dec 15 08:26:29 rautu rtpengine[10473]: [dgp00b8p8q45f0vba38o] Creating new call Dec 15 08:26:29 rautu rtpengine[10473]: [dgp00b8p8q45f0vba38o] Opened ports 50110..50111 for media relay Dec 15 08:26:29 rautu rtpengine[10473]: [dgp00b8p8q45f0vba38o] Opened ports 50118..50119 for media relay Dec 15 08:26:29 rautu rtpengine[10473]: [dgp00b8p8q45f0vba38o] Returning to SIP proxy: d3:sdp717:v=0#015#012o=Mozilla-SIPUA-34.0 24971 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-lite#015#012m=audio 50110 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendonly#015#012a=rtcp:50111#015#012a=rtcp-mux#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:vSMhw6pr2O+DWrL+7O1chSiCrSZ5YJW4esLlgNv/#015#012a=setup:actpass#015#012a=fingerprint:sha-1 42:C4:38:DC: ... Dec 15 08:26:29 rautu rtpengine[10473]: [dgp00b8p8q45f0vba38o] ... A6:4A:A6:70:49:6A:96:81:BE:9A:17:7D:FE:02:00:2C#015#012a=ice-ufrag:GL0QSUR3#015#012a=ice-pwd:X1irWG0YUiMpcTLgHo9OHgNl1NbT#015#012a=candidate:Wz1BLNSFenkyID4v 1 UDP 2130706431 0.0.0.0 50110 typ host#015#012a=candidate:Wz1BLNSFenkyID4v 2 UDP 2130706430 0.0.0.0 50111 typ host#015#0126:result2:oke

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Change Invite message to 302 Moved
by Leky Shakur 14 Dec '14

14 Dec '14

Hello, I have a situation, where provider does not understand reivnites when user makes a Call Forwarding via Asterisk server, it only understands 302 Moved messages in CF situations (or in Call tranfer). So my scenario is like this: Provider sends Invite to Kamailio, I forward it to Asterisk server. where a user make a call forwarding to Kamailio (INVITE message), who then forwards it to provider - my question if, can I change the INVITE that Asterisk send Kamailio server and change it to 302 Moved message and relay it to provider (I can detect when there is an CF, because Asterisk sends diversion header in the INVITE towards Kamailio)?

1 0

t_suspend/t_continue
by Ahmed Salem 13 Dec '14

13 Dec '14

Hi, I have 2 kamailio servers behind a load balancer. How can I use t_suspend on one server and t_continue on the other one. I already have the index and label of the transaction saved in a common database shared by the two kamailio servers. The reason I want to do this is for push notifications for iOS and Android, and I can't guarantee that the caller and the callee will register on the same server because of the load balancer. Thanks Ahmed

2 5

Error When Configuring the sipcapture module.
by Mahmoud Ramadan Ali 13 Dec '14

13 Dec '14

Hi Dears, I'm trying to configure the SIP Capture module with Kamailio BUT i'm receiving that Error. Dec 13 09:20:53 debian /usr/local/sbin/kamailio[5183]: ERROR: sipcapture [hep.c:84]: hep_msg_received(): ERROR: sipcapture:hep_msg_received: not supported version or bad length: v:[82] l:[69] Any ideas how to solve this please ?! Best Regards and Thanks in advance

1 0

kamailio for newbies!
by Aria Mill 13 Dec '14

13 Dec '14

Hello, Kamailio is difficult. Many of us want to use it because is open source and it's flexible. but to tell you the truth after 3-4 hours of playing with it I am frustrated, and I am starting to hate it! Why has nobody made a kamailio video series on youtube?-example of how to install kamailio-example of how to admistrer kamailio, and what other programs you need to work with;-example how to create users in Kamailio for enterprise use.-example of running kamailio outside enterprise like a SBC with a public ip;-example of how can you run kamailio as a load balance server.-example how to save money on your sip infrastructure using kamailio instead of cisco cube for example or sonus..or any other example you might thing is useful in a sip environment! I look at youtube and I see Kamailio this and kamailio that, but no actual tutorial! This is 2014, people don’t waste time reading documentation, youtube is much more efficient...I think Kamailio will not have the success of it's potential until senior community members will not take the plunge and broadcast their knowledge on a proper video channel! Asterisk has every scenario up there based on these info I was able to setup in less that 3 days my office and 40 people are making calls......It's unbelievable that after so many years nobody has made a godam explicit tutorial for newbies in kamailio...I am so frustrated right now!!

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Create and control a new dialog to different server during main dialog
by Konstantin M. 12 Dec '14

12 Dec '14

Hello, I want to achieve the following schema: 1. Accept an initial invite 2. Create separate dialog to different SIP server (probably with some headers from point 1) 3. Wait until that different SIP server will complete dialog. 4. Based on result of point 3, transfer invite from 1 to the next steps or decline a call. The problem I've experienced - how to create a separate dialog and control it inside main dialog? Thanks a lot!

2 2

Planning to switch committing to github
by Daniel-Constantin Mierla 11 Dec '14

11 Dec '14

Hello, I setup the mirror of Kamailio github repository to kamailio.org, can be seen online at: - http://git.kamailio.org/gitlist/index.php/kamailio/ Mirroring is triggered by a push to github, so it is pretty much in real time. Right now the commit and mirroring flow is: - commit to git.sip-router.org triggers mirroring (push) to github, which triggers a mirroring (fetch) on git.kamailio.org Practically with a commit pushed directly to github, the first part of the above is skipped. All seems to be ok, my last commits were done to test the mirroring including the notification emails - I will send a separate message about what we can get there. The missing parts are the deployment of gitweb in order to have the links from archived email notifications still valid and pointing git.sip-router.org to kamailio mirror server. Considering all above, I propose to stop committing to git.sip-router.org and pushing to github starting either next Monday (Dec 15) (or Wednesday (Dec 17) if not ready by Monday). Existing developers have to send me the github username to add it to kamailio project. Any suggestions and comments are welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 4

Charging System
by Abdul Gafar 11 Dec '14

11 Dec '14

Hi all I'm trying install IMS charging module,is this posible use kamailio server, without install other module IMS ? //Gafar

2 2

Appending/removing a value associated with domain
by Olli Heiskanen 11 Dec '14

11 Dec '14

Hello, Something I've been wondering about meddling with sip uris with Kamailio: I know it's possible to translate between a number prefix and a domain using PDT, but is this possible: Having a numeric or alphanumeric value stored in db, associated to a domain and appended to / removed from a number during processing a sip message? For example 112222333(a)mydomain.com --> 333.5555(a)mydomain.com where value 5555 and mydomain.com are associated to one another in db somewhere. In this example I could have a pdt with prefix 112222 and domain mydomain.com, but the value 5555 should be stored somewhere as well. So one part is to find a db table suitable for storing a value and associating it with a domain, and the other part is to change the sip uris. I'll appreciate any comments! cheers, Olli

1 0

Federation Kamailio
by nadie 11 Dec '14

11 Dec '14

Hello. I installed kamailio. All OK. But when i call to other servers. ostel.co. for example. no ring. What element permits federation between kamailios? DNS Records? Thank you.

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sr-users December 2014