sr-users September 2010

sr-users@lists.kamailio.org

58 participants
97 discussions

Avoid resolving names by DNS
by Santiago Soares 02 Sep '10

02 Sep '10

Hello, I'm using kamailio as a simple load balancer. My dispatcher list is something like that: 1 sip:server1:5060 1 sip:server2:5060 2 sip:server1:5060 and /etc/hosts is 10.1.1.1 server1 10.1.1.2 server2 When I want to do some maintenance task in one of the servers, server2, for exemple, I simply change /etc/hosts like that: 10.1.1.1 server1 10.1.1.1 server2 And all the traffic is redirected to server 1. It's a simple yet effective way to remove a server from balancing. … [View More]

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Re: [SR-Users] [sr-dev] Table version problems
by marius zbihlei 02 Sep '10

02 Sep '10

Raúl Alexis Betancor Santana wrote: > Hi ... jus trying Kam 3.0.3 ... I get blocked on this error: > > ERROR: <core> [db.c:408]: invalid version 0 for table 0 found, expected 6 > (check table structure and table "version") > ERROR: auth_db [authdb_mod.c:250]: error during table version check. > > It should be checking agains subscriber table version ... but I don't know why > it try to found '0' table ... > > if I do a: > > insert into version (… [View More]

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topoh and SDP endpoints
by Alex Balashov 01 Sep '10

01 Sep '10

The topoh module doesn't touch IP-identifying SDP information, does it? Has there been any interest in implementing that down the road? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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help to use force_rtp_proxy([flags [, ip_address]]).
by peter_green lion 01 Sep '10

01 Sep '10

hi all, i am a new user kamailio. i have configure kamailio with RTP proxy, but i have a problem in using : force_rtp_proxy("c","192.168.1.10")because i want to change to force to this ip. my configure is : when server receive "200 OK" it change value in "c= IN IP4 <ip rtp server>" to "c = IN IP4 192.168.1.10" i configure as : kamailio.cfg : #!ifdef WITH_NAT if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy("c","192.… [View More]168.1.10"); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } when i make call from sip a to sip b, sip b answer .trace as : U 115.78.129.190:54337 -> <server ip>:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP <server ip>;branch=z9hG4bK00c.daa8fe8.0. Via: SIP/2.0/UDP 192.168.1.10:50937;received=115.78.129.190;branch=z9hG4bK-d8754z-1f66a816d651d504-1---d8754z-;rport=63930. Record-Route: <sip:<server ip>;lr;nat=yes>. Contact: <sip:102@192.168.1.10:8576;rinstance=8392ffb3fe461110>. To: <sip:102@<server ip>:5060>;tag=d179a842. From: <sip:101@<server ip>:5060>;tag=1c61b708. Call-ID: ZTQzYjZkYzFjODE1MWFlNjIwNmQ2ZGU5MWUxYmM1NjQ.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: PortGo v6.0, Build 07282010. Content-Length: 237. . v=0. o=- 30452887 30452887 IN IP4 169.254.202.160. s=http://www.portsip.com. c=IN IP4 169.254.202.160. t=0 0. m=audio 21480 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U <server ip>:5060 -> 115.78.129.190:63930 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.10:50937;received=115.78.129.190;branch=z9hG4bK-d8754z-1f66a816d651d504-1---d8754z-;rport=63930. Record-Route: <sip:server ip;lr;nat=yes>. Contact: <sip:102@115.78.129.190:54337;rinstance=8392ffb3fe461110>. To: <sip:102@server ip>:5060>;tag=d179a842. From: <sip:101@<server ip>:5060>;tag=1c61b708. Call-ID: ZTQzYjZkYzFjODE1MWFlNjIwNmQ2ZGU5MWUxYmM1NjQ.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: PortGo v6.0, Build 07282010. Content-Length: 237. . v=0. o=- 30452887 30452887 IN IP4 169.254.202.160. s=http://www.portsip.com. c=IN IP4 169.254.202.160. t=0 0. m=audio 21480 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. thanks for help me. regards. beter_green [View Less]

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Cluecon 2010 presentation
by Daniel-Constantin Mierla 01 Sep '10

01 Sep '10

Hello, for those interested, I uploaded my presentation at Cluecon on the web site, link: http://bit.ly/cFDNcX My one and a half day presence in Chicago was surprising good for kamailio and ser - people started to get the real value of the project and catch up with latest features. Moreover, the first day had a lot of presentation with kamailio and ser (see agenda at: http://www.cluecon.com/schedule): - Phillip Zimmermann - did his demo with iptel.org sip service - Mahesh Paolini-… [View More]

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ratelimit module migration from (k) to generic
by Ovidiu Sas 01 Sep '10

01 Sep '10

Hello all, The (k) ratelimit module is an enhanced version of the original (s) ratelimit module. Are there any objections in moving the ratelimit module from (k) to generic and purging the (s) version? Regards, Ovidiu Sas

2 3

help to get ip from contact header in INVITE sip message.
by peter_green lion 01 Sep '10

01 Sep '10

hi all, please help me to get ip add from Contact header in INVITE message. i see that : ct =<sip:101@192.168.1.10:63027;rinstance=85f97591c218086a> have the ip or domain ("192.168.1.10"). so please suggest to get this ip. thanks so much. beter green

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sr-users September 2010