sr-users October 2011

sr-users@lists.kamailio.org

69 participants
117 discussions

by Lello

Hi, I'm working with a softphone project named mjSip, now I've implemented the TLS protocol in the softphone and I would try it with 2 iptel account setted on two pc, on the first I have: # MjSip UA configuration file # ___________________________________________ # host_port=5061 transport_protocols= tls outbound_proxy=sip.iptel.org from_url=<sip:alexlel@iptel.org> username=alexlel realm=iptel.org passwd=2387lelloh debug_level=8 log_path=log max_logsize=2000 do_register=yes keepalive_time=8000 _______________________________ the second is such as this. However I have no response in SSL/TLS by the iptel server. Does it work in TLS? Thanks, Regards.

12 years, 7 months

Employment you've been searching!

by apache＠iptel.org

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12 years, 7 months

Can't create a new account with kamctl

by Peter Schrock

I have finally got to the section of the install section "10. Ready to Rock". Unfortunately, I did get the error message "ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN". I did the two steps that the instructions gave and cannot find ".kamctlrc" to "add SIP_DOMAIN=mysipserver.com" Please help. I am not quite understanding what should be done to fix this Peter

12 years, 7 months

Kamailio Developer Seminar, Oct 24-24, San Francisco

by Daniel-Constantin Mierla

Hello, the place and date for Kamailio/SER Developer Seminar have been set: San Mateo (just south of San Francisco, CA, in the nearby of international airport), for one day and a half, Oct 24-25, 2011. I posted more details, including about content and registration, at: * http://www.kamailio.org/w/2011/10/kamailio-developer-seminar-san-francisco-… Participating fee is 250USD. Selection of attendees is done by the rule of first come first served. The number of seats is limited, don't wait too long if you want to attend. Looking forward to meeting many of you there. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

12 years, 7 months

Can anyone help for a small amount of money.

by Henrik Aagaard Sørensen

Hi everyone. I'm still facing problems with my Kamailio Dispatcher <-> FreeSWITCH setup. Everything works when connecting directly to FreeSWITCH. When connecting to Kamailio, which should work as a dispatcher (load-balancer) in front of FreeSWITCH I get "Too many hops", "Timeouts" etc. My test setup with SIP clients is an iPhone, an Android, an iPad and a laptops with Ubuntu. Can someone help me out, for a small amount of money, until I can get it to work? Hopefully it's just the kamailio.cfg which needs to do things correctly. Currently I'm using this http://pastebin.com/mahKECAw with help from Morten in this mailing list.

12 years, 7 months

Creating a database error

by Peter Schrock

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 not found ERROR: Creating core database and grant privileges failed! I looked at the files and cannot find the problem. Please help. Peter

12 years, 7 months

Kamailio Dispatcher and FreeSwitch, Too many hops.

by Henrik Aagaard Sørensen

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 between clients they just send the packets between each other. Can anyone help me out? I've tried to Google a lot for this problem and asked in several IRC channels, mailing lists and forums. Without any luck.

12 years, 7 months

Re: [SR-Users] Problem with Route containing two elements

by davy van de moere

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 situation ignore the first uri in the route header? Or am I still not getting it :) ? thx ! BYE sip:05xxxxxx at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>: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 at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> ;lr=on;ftag=as5716fec8> Max-Forwards: 69 From: "Some_Telecom" <sip:02xxxxxxxx at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>;tag=as5716fec8 To: <sip:05xxxxxx at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6 at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060;transport=udp X-Enswitch-Source: their_internals:5060

12 years, 7 months

proxy BYE to itself

by caio

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 same IP (eth iface) address, but one at port 5080, and the proxy at 5060 (udp). The configuration is based on kamailio.cfg file provided by the v3.1.5 (with a few modules added, like dialplan and cr..etc). If A is who send the BYE, everything goes well. I attach the full sip capture. Thanks for you time. Claudio

12 years, 7 months

Problem with Route containing two elements

by davy van de moere

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 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

12 years, 7 months

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sr-users October 2011