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20 participants
37491 discussions

Creating a database error
by Peter Schrock 12 Oct '11

12 Oct '11

This is what I am getting: INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command … [View More]

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Kamailio Dispatcher and FreeSwitch, Too many hops.
by Henrik Aagaard Sørensen 12 Oct '11

12 Oct '11

I have a setup with Kamailio as dispatcher in front of a FreeSwitch server. This is my kamailio.cfg: http://pastebin.com/8PR2GFBD I'm currently getting "Too many hops" when calling between SIP clients. I am able to call to FreeSwitch and listen to voicemail, hold music etc. After a long conversation with a FreeSwitch expert, and some tests, I was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch. I've also run tshark on both FreeSwitch and Kamailio and when calling … [View More]

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Re: [SR-Users] Problem with Route containing two elements
by davy van de moere 12 Oct '11

12 Oct '11

Owki, reading RFC-3261 again ;) Imho Kamailio does what it is supposed to do, it forwards the packet to the first element in the Route header. But as I received the BYE packet from <sip:their_sbc> , that one should have stripped itself out of the Route header, making everything just work, right? Now I can hackishly fix it with a rewritehostport in the middle. But that will eventually not scale up that nicely. Ergo, does anyone have a good idea on how to e.g. have Kamailio in this … [View More]

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proxy BYE to itself
by caio 12 Oct '11

12 Oct '11

Hello, I have the following issue. A calls B, talk for a while, then B sends a BYE, but A never receives it. The BYE is transmited from the proxy listening interface:port to itself, but not to the asterisk port. U 2011/10/08 00:29:29.207335 Kamailio_IP:5060 -> Kamailio_IP:5060 BYE sip:Kamailio_IP;lr=on SIP/2.0. The scenario is: A (pstn subs) --> Asterisk:5080 --> Kamailio:5060 --> B (registered sip ua) Asterisk and Kamailio are in the same box, listening both at the … [View More]

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Problem with Route containing two elements
by davy van de moere 11 Oct '11

11 Oct '11

I'm trying to integrate to an integrics enswitch, almost everything works like a charm, except on BYE packets as Kamailio in my setup forwards these incorrectly. Digging somewhat I believe the culprit sits in the Route header which comes from the enswitch: Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Kamailio takes into account the first argument (the thing sip:their_sbc...) on which it cannot … [View More]find the correct transaction. And forwards it back to the sender, instead of forwarding it correctly to my freeswitch. which gives a storm of traffic ;) If I compare that with an ACK packet in the same transaction , having this as a Route header : Route: <sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Kamailio sends it along correctly. IMHO if I could convince Kamailio to always take the last part of the Route header into account and ignore the first one it would correctly work. What would be a good approach? Rewriting the route header looks abit harsch. disclaimer : I'm not sure which component is disrespecting the RFC, if any, or if I just fully not get something :) Short overview + packet detail (with some obfuscation): My setup: their_sbc --- my_kamailio --- freeswitch --- my_kamailio --- enduser BYE sip:05xxxxxx@my_kamailio:5060;transport=udp SIP/2.0 Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP their_sbc;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Max-Forwards: 69 From: "Some_Telecom" <sip:02xxxxxxxx@their_sbc>;tag=as5716fec8 To: <sip:05xxxxxx@their_sbc:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6@their_sbc CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx@my_kamailio:5060;transport=udp X-Enswitch-Source: their_internals:5060 BYE sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio;lr=on;ftag=as5716fec8 SIP/2.0 Record-Route: <sip:05xxxxxx@my_kamailio;lr=on;ftag=as5716fec8> Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP my_kamailio;branch=z9hG4bK1ae8.b4f95af2.0 Via: SIP/2.0/UDP my_kamailio;rport=5060;branch=z9hG4bK1ae8.a4f95af2.0 Via: SIP/2.0/UDP their_sbc;rport=5060;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Max-Forwards: 67 From: "Some_Telecom" <sip:02xxxxxxxx@their_sbc>;tag=as5716fec8 To: <sip:05xxxxxx@their_sbc:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6@their_sbc CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx@my_kamailio:5060;transport=udp X-Enswitch-Source: their_internals:5060 [View Less]

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[OT] SIP on the Web
by Iñaki Baz Castillo 11 Oct '11

11 Oct '11

Hi folks, I would like to show a project my colleage José Luis Millán and me are working on. It's about SIP protocol running in a web browser. When RTCweb (media capable web browsers) becomes a reality both technologies together will allow real SIP endpoints coded in JavaScript running in a web browser and interoperating with real SIP networks. Bye bye Flash and custom protocol gateways. http://sip-on-the-web.aliax.net/ Cheers. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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limiting concurrent calls with dialog module
by Uri Shacked 11 Oct '11

11 Oct '11

Hi, I am trying to create concurrent calls limitations. two questions: 1. should i use dialog module? or is there a better module? 2. i was trying to work with the dialog module, but didnt understand how and where to configure the commands and which to use.... any examples or ideas? BR, Uri

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releasing v3.2.0
by Daniel-Constantin Mierla 11 Oct '11

11 Oct '11

Hello, it is time to set the time to release v3.2.0. From my point of view, testing went very well, considering also that core and main modules were not touched much during this devel cycle, the most of the efforts were on new modules and application service components such as presence server, Lua ... A convenient date for me will be Tuesday, October 18, 2011, allowing to work a bit on packaging and docs. If there are other preferences, jump in this discussion. Cheers, Daniel -- … [View More]

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(no subject)
by alan chiu 10 Oct '11

10 Oct '11

http://toehugger.com/CRE/info.php?html143

1 0

Audio quality issue
by Austin Einter 10 Oct '11

10 Oct '11

Hi I am using Kamailio 3.1.5 . I am using RTP proxy also. I have used default kamailio.cfg.sample fiile , and just added line #!define WITH_NAT. I have another Main proxy. I wanted all my signalling and media packets should just pass through machine where Kamailio and RTP proxy are running. With this I found, call is established, all signalling and media packets are passing through kamailio / rtp-proxy. So far so good. One way audio stream (from called party to calling party) quality is good.… [View More]

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