sr-users September 2010

sr-users@lists.kamailio.org

58 participants
98 discussions

Using multiple media proxy pools
by Stagg Shelton 02 Sep '10

02 Sep '10

I have a requirement that I hope can be accomplished. I have to connect to three providers who will be delivering a private IP Layer 2 ethernet endpoint. I need to be able to direct the SIP calls via Kamailio and have the RTP media process through mediaproxy. I also need to try and avoid using separate physical media proxy machines for each SIP provider. Is it possible to run multiple mediaproxy dispatchers bound to separate IP addresses, and run multiple relays with IP addresses corresponding to the various dispatcher subnets. Basically I want to identify pools of mediaproxy servers and choose them based on the provider origination or termination. Alternatively we could use RTP Proxy, though we do not have as much experience with RTP Proxy. Any suggestions that you may have is appreciated. Thanks Stagg

2 1

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. The thing is that it works on kamailio 1.5.3, but on kamailio 3.0.3 it doesn't work, since kamailio appears to resolv names only by DNS. I receive this message on startup: Sep 1 12:04:32 install /usr/local/sbin/kamailio[16077]: ERROR: dispatcher [dispatch.c:241]: could not resolve server1 Sep 1 12:04:32 install /usr/local/sbin/kamailio[16077]: ERROR: dispatcher [dispatcher.c:247]: no dispatching list loaded from file Sep 1 12:04:32 install /usr/local/sbin/kamailio[16077]: ERROR: <core> [sr_module.c:874]: init_mod(): Error while initializing module dispatcher Is there any way to avoid this behavior, and force kamailio to use /etc/hosts? Thank you. Santiago Soares Fone: (41) 8488-0537

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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 (table_name,table_version) values('0','6'); > > It get solved ... but thats only a trick. > > Hello, The check is done once (at fixup time) for the www_authorize and proxy_authorize calls. The table name is the second parameter given to those calls. Be sure to use www_authorize("explample.com", "subscriber"); Cheers Marius

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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/

2 2

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","192.168.1.10"); } 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

2 1

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-Subramanya - a clould based system for telephony - Scott Burkett - back-end platform for web telephony - Robin Rodrigues - several nice examples using upcoming 3.1 addition: embedded http server - Irv Shapiro - ifbyphone platform - programmable telephony applications - I ended the list (the link above), no much left for me to say :-) , but I enjoyed chatting before a lot with folks in the breaks Hopefully these presentations will be soon an event's site. I will try to publish more about each one, they were interesting use cases. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

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