sr-users March 2009

sr-users@lists.kamailio.org

92 participants
149 discussions

by Bruno Rodrigues

Hi All, First, sorry I have a bad english. Second, Anybody know if is possible redirect the call to one MediaServer when the destination answer the call? In the Media Server, the callee will hear a announcement, and the destination will hear a another announcement. The announcement has the same duration. When the announcement finish, the RTP will be established between the UA1 and UA2 The signaling I think will be this UA1 Kamailio UA2 Media Server Invite -----------------> - Received a normal Invite Invite----------------> - Forward The Invite <------------------200 - Receive the 200 OK <------------------200 - Forward the 200 OK <-------------re-Invite re-Invite ------------> - Kamailio will update the media part to the Media Server <----RTP Announce 1-----> - RTP are established <----------------------RTP Announce 2 ---------------------------------> - RTP are established <-------------re-Invite re-Invite ------------> - Kamailio after the announcement are finished update the Media part to UA1 ip media and UA2 ip media <-----------------RTP-------------------------> - RTP between UA1 and UA2 If have another way to do this, please let me know Thank You in advanced Bruno Rodrigues

15 years, 8 months

[Kamailio-Users] DNS lookup inside cfg

by Uriel Rozenbaum

Genlemen, I'll be quick: is there any way to perform a DNS lookup within the .cfg? The example would be: avp(s:server_name) = "kamailio.org"; avp(s:server_ip) = "192.168.1.100"; if (DNS_LOOKUP_FUNCTION(avp(s:server_name)) == avp(s:server_ip)) { ... do something ... } else { ... do something else ... } Thanks in advance, Uriel

15 years, 8 months

[Kamailio-Users] [OT] sip router masterclass

by Daniel-Constantin Mierla

Hello, my apologizes if this email look business oriented for this list, but to answer some questions in the last time, there is now a dedicated professional training program for kamailio (openser) and sip-router.org: http://www.asipto.com/index.php/sip-router-masterclass/ Community wise, each time will be a Social Networking Event (dinner) where everybody can join and have a chat with other folks developing or using these projects. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

15 years, 8 months

[Kamailio-Users] new module: memcached

by Henning Westerholt

Hi all, i just added a new module to the repository - 'memcached'. Its allows to connect to a memcached server, and supports add, get, delete and atomic operations via pseudo-variables. Further details can be found in the module documentation: http://kamailio.org/docs/modules/devel/memcached.html If the pv module is imported into the sip-router it should be also work there, i'll migrate the module then to this tree. Best regards, Henning Westerholt

15 years, 8 months

[Kamailio-Users] Strange behaviour when RADIUS server is down

by Antonio Reale

Hi all, I see a very strange behaviour using accounting on RADIUS. If RADIUS server is up and running, all is OK. If it is down dialog module (in db mode) can't work properly: In the log I see: after 200 OK to INVITE CRITICAL:dialog:log_next_state_dlg: bogus event 6 in state 2 for dlg 0xb61c0918 [360:515907930] with clid 'B8108B3CEBD9B8E3E1CC23DE8361D35F@sipsvr' and tags 'C319BA6F3E777E04867202774CE6443B' '' after BYE CRITICAL:dialog:log_next_state_dlg: bogus event 7 in state 2 for dlg 0xb61c0918 [360:515907930] with clid 'B8108B3CEBD9B8E3E1CC23DE8361D35F@sipsvr' and tags 'C319BA6F3E777E04867202774CE6443B' '' and no record is written in the dialog table. All come back to work after starting RADIUS server. This is not the main problem... In some circumstance kamailio fails to route properly the calls, due retransmissions; for instance after the client answers I see a lot of 200 OK sent to kamailio because it doesn't send the ACK immediately. In the log I see for each 200 OK: ... Mar 9 11:34:01 [27466] DBG:tm:timer_routine: timer routine:3,tl=0xb61a8ecc next=(nil), timeout=75 Mar 9 11:34:01 [27466] DBG:tm:delete_handler: removing 0xb61a8e68 Mar 9 11:34:01 [27466] DBG:tm:delete_cell: delete_cell 0xb61a8e68: can't delete -- still reffed (1) Mar 9 11:34:01 [27466] DBG:tm:set_timer: relative timeout is 2 Mar 9 11:34:01 [27466] DBG:tm:insert_timer_unsafe: [3]: 0xb61a8ecc (77) Mar 9 11:34:01 [27466] DBG:tm:delete_handler: done ... I was not able to replicate this problem. Solved after starting RADIUS server, but the problem doesn't occur if I stop it. I'm using kamailio 1.4.3, but the last problem occurred also on a production system with openser 1.2.3. After customer's RADIUS server went down, openser wasn't able to route all calls. Thank you very much for support. Regards, Antonio.

15 years, 8 months

[Kamailio-Users] sl README does not know send_reply()

by jh＠tutpro.com

the new send_reply() function is not documented in sl/README. another related thing: would it be logical to also have corresponding reply_error() function especially when t_relay seems to create transaction even in case of some internal errors. -- juha

15 years, 8 months

[Kamailio-Users] Problem starting Kamailio

by Sara EL KABIRI

Hello, I've installed kamailio 1.4.3 (without tls) on ubuntu by following the provided INSTALL guide. I want to use it as a presence server but i've got some problems starting it. "kamctl start" returns nothing. and "kamctl moni" returns "Error opening kamailio's FIFO /tmp/kamailio_fifo" I've checked in my .cfg file and the 'loadmodule' and 'modparam' lines related to mi-fifo module are uncommented. I've checked also the mi-fifo module was installed indeed (mi_fifo.so appears in /usr/local/lib/kamailio/modules directory). Could anyone help please? -- Sara

15 years, 8 months

[Kamailio-Users] Force rtp proxy

by michel freiha

Hi all, I have a question regarding rtp proxy...I need to force rtp proxy in order that all rtp packets pass through...The issue now is the first INVITE does contains the PSTN GW IP instead of RTP Proxy IP...My config is as below: if (is_method("INVITE")) { # setflag(4); # do accounting fix_nated_sdp("1"); } if(!cr_route("default", "0", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In cas of failure, re-route the request t_on_failure("1"); t_on_reply("1"); force_rtp_proxy(); t_relay(); } onreply_route[1] { # if (t_check_status("2[0-9][0-9]") ) if((t_check_status("200|183|180") && search("Content_Type: application/sdp")) || search("Content-Type: application/sdp")) { force_rtp_proxy(); } exit; } Please let me know what's wrong here and how i can fix this issue Regards

15 years, 8 months

[Kamailio-Users] Forgetting to call unforce_rtp_proxy

by mayamatakeshi

Hello, is there any risk of having resources leaking if I fail to call unforce_rtp_proxy? Like having rtpproxy not freeing ports? I'm mean, my cfg is becoming bigger and I may fail to call this function somewhere. Should I be extremely careful about this? regards, takeshi

15 years, 8 months

[Kamailio-Users] LCR With 300,000 Routes

by Geoffrey Mina

I have spec'd out how my LCR table would look based my current PSTN gateway providers. Looks like I would need to load roughly 300,000 routes to properly represent everything. This server needs to be able to support spikes of 100 call setups per second. What is the general consensus on this? And, if it's a No-Go, what alternatives do I have? I have looked briefly at the CarrierRoute module, but haven't actually played with it. Thanks! Geoff

15 years, 8 months

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sr-users March 2009