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42 participants
37519 discussions

Best practise for limiting concurrent calls across a cluster
by jay binks 08 Sep '15

08 Sep '15

Hey all, So I have a cluster of Kamailio servers ( 4 servers currently, soon to be 8 ), I'm looking for suggestions about the BEST way to achieve concurrent call limiting on a per customer basis, across the whole cluster. Initially I mis-read the dialog module documentation and assumed that dialog would provide me this ability, when used with a database. however it seems that the dialog module does not pull data from the DB after the initial startup. I know I can use sql ops to increment and decrement using event_route[dialog:start] and event_route[dialog:end]. however the database I've chosen ( for other valid reasons ) does not have an atomic increment and decrement. I could add yet another DB, but that just adds more failure points. so lets forget my setup, Im wanting suggestions about the BEST setup for this sort of thing. while remaining fault tolerant, and preferably without relying on any single point of failure. Sincerely Jay

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core parameter onsend_route_reply=1 doesn't execute onsend_route block for received replies
by Julia Boudniatsky 07 Sep '15

07 Sep '15

Hello, I set some logs in onsend_route and it works fine for requests. With setting core parameter onsend_route_reply=1 ( yes, on), i expect to see logs for replays also, but it not occurs. Is this a bug or I miss something? version: kamailio 4.2.6 (x86_64/linux) db77ac BR, Julia

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Siremis 4.2 with PHP 5.6
by Patrick Bezensiertblubb 07 Sep '15

07 Sep '15

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Kamailio + Rtpengine < - > Freeswitch + ZRTP
by Alexandru Covalschi 07 Sep '15

07 Sep '15

Hello! Was trying to perfrom subj interconnection via http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbс <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> via that guide. Kamailio and FS are on same host on Amazon EC2. The issue is - when I have proxy_media=true on FS, ZRTP is OK, but the leg_b user can't hear anything (but his voice is transmitted OK). When I set proxy-media to false the voice is OK, but something is with ZRTP so Jitsi can't establish ZRTP handshake. When I set bypass-media=true on FS both ZRTP and voice established, but quality is sometimes bad and the delay is too big. Anyone has any ideas? Kamailio 4.3, FS 1.4 <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

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Thoughts on setup with Kamailio and FreeSWITCH
by Michael Nielsen 07 Sep '15

07 Sep '15

Hello everyone... I'm testing various ways of setting up Kamailio and FreeSWITCH (as VAS-handler). I'm struggling a bit on what Kamailio should do and FreeSWITCH should do. My basic setup is to handle voicemails, internal calls between subscribers and outgoing calls to a SIP gateway to GSM. My idea is to have Kamailio handle all calls and routing between subscribers and the SIP gateway and only have FreeSWITCH handle voicemail (to save the audio file and e-mail it). How does that sound? Or should I route everything to FreeSWITCH via Kamailio and have FreeSWITCH route calls between subcsribers and SIP gateway as well? What are best practises etc...?

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Question about in call both side redirection
by Андрей Ярин 07 Sep '15

07 Sep '15

Hello (sorry for my bad english) - i try to create voice record service by request. User A call to user B. In call by pressing combination like *55 Kamailio must redirect both sides to asterisk, whitch create dynamic conference room with recording. As i understand i need to use dlg_refer() from dialog module, but in log file i get: Konsole output Sep 4 10:45:14 voipc-node2 /usr/local/sbin/kamailio[18941]: ERROR: dialog [dlg_req_within.c:85]: build_dlg_t(): no contact available Sep 4 10:45:14 voipc-node2 /usr/local/sbin/kamailio[18941]: ERROR: dialog [dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t In script i try to refer with: dlg_refer("callee", "sip:100@10.10.9.209"); dlg_refer("caller", "sip:100@10.10.9.209"); -- Ярин Андрей Инженер ООО "И.С.-Телеком" 8(351)7786878 доб 113 8(351)2555724 доб 113 8(351)2555787 доб 113

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Route all traffic from Kamailio to FreeSWITCH after inital checks...
by Michael Nielsen 07 Sep '15

07 Sep '15

I'm running the standard Debian 7 Kamailio 4.3 package. I've added the following to my kamailio.cfg: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS #!define WITH_ANTIFLOOD After authentication, nat etc. is done is there an easy way to transfer everything else to FreeSWITCH? I've looked at the dispatcher module, but don't know where to put the "transfer" in kamailio.cfg?

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Re: [SR-Users] webrtc
by Slava Bendersky 07 Sep '15

07 Sep '15

Hello Giovanni, I fixed webrtc call working right now. I checked debug again and I so 408 error. There was missing record_route() for first INVITE. Thank you for help. Slava. From: "Giovanni Maruzzelli" <gmaruzz(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 5 September, 2015 07:50:29 Subject: Re: [SR-Users] webrtc First thing, check that you can have a webRTC call directly with FreeSWITCH, without passing by Kamailio. If that works, and it does not work through Kamailio, then provide kamailio config, freeswitch config, and debug log from both. On Sat, Sep 5, 2015 at 12:51 AM, Slava Bendersky < volga629(a)skillsearch.ca > wrote: Hello Everyone, I am trying setup webrtc call with freeswitch. [root@canlvprx01 kamailio]# rpm -qa | grep kamailio kamailio-4.3.1-4.4.fc21.x86_64 Case: INTERNET eth0 | eth1 LAN | Web Browser ---->webrtc ----> kamailio ----> freeswitch Getting this error Sep 4 00:22:06 canlvprx01 /usr/sbin/kamailio[11820]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Sep 4 00:22:10 canlvprx01 /usr/sbin/kamailio[11817]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Sep 4 00:22:14 canlvprx01 /usr/sbin/kamailio[11818]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Tried check debug don't see any thing special. Any help thank you. Please let me know which logs will good to provide. Slava. _______________________________________________ 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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _______________________________________________ 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

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Re: [SR-Users] webrtc
by Slava Bendersky 06 Sep '15

06 Sep '15

Hello Giovanni, Here SDP webrtc call traffic capture on kamailio side http://fpaste.org/264289/15798761/ password from paste: kamailio Slava. From: "Giovanni Maruzzelli" <gmaruzz(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 5 September, 2015 07:50:29 Subject: Re: [SR-Users] webrtc First thing, check that you can have a webRTC call directly with FreeSWITCH, without passing by Kamailio. If that works, and it does not work through Kamailio, then provide kamailio config, freeswitch config, and debug log from both. On Sat, Sep 5, 2015 at 12:51 AM, Slava Bendersky < volga629(a)skillsearch.ca > wrote: Hello Everyone, I am trying setup webrtc call with freeswitch. [root@canlvprx01 kamailio]# rpm -qa | grep kamailio kamailio-4.3.1-4.4.fc21.x86_64 Case: INTERNET eth0 | eth1 LAN | Web Browser ---->webrtc ----> kamailio ----> freeswitch Getting this error Sep 4 00:22:06 canlvprx01 /usr/sbin/kamailio[11820]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Sep 4 00:22:10 canlvprx01 /usr/sbin/kamailio[11817]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Sep 4 00:22:14 canlvprx01 /usr/sbin/kamailio[11818]: WARNING: <core> [forward.h:164]: msg_send(): TCP/TLS connection for WebSocket could not be found Tried check debug don't see any thing special. Any help thank you. Please let me know which logs will good to provide. Slava. _______________________________________________ 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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _______________________________________________ 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

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Topoh configuration question
by Asgaroth 06 Sep '15

06 Sep '15

Hi all, I have a quick question with regards the topoh module, is it possible to have the topoh module only operate on a particular interface and direction. I have some edge proxies and I am trying out the topoh module, what I see happen is that messages originating from behind/inside the edge proxy is topoh'd out, which is exactly what I was looking for, but it appears to do the same for messages originating from the wild, so messages pushed in behind/inside the proxy is also encoded, which is not what I was expecting. Is it possible to configure the topoh module to activate for messages being sent out of the edge proxy into the "wild", but not the other way around, or, decode if it finds an encoded messages who's destination is "behind" the proxy. Any thoughts are much appreciated. Thanks

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