- sr-users - kamailio.org

Re: [SR-Users] Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

That works. Thanks. From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Thursday, June 02, 2011 4:28 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gary Chen Subject: Re: [SR-Users] Problem using dispatcher. On 6/2/11 9:41 PM, Gary Chen wrote: Thanks for the quick reply. How can I fix this? When you set t_on_failure for an INVITE, the respective failure route will be executed for any reply >=0. What you have to do is to not re-route to a new destination from failure route, simply exit for those reply codes, like: if(t_check_status("401|407")) exit; You don't handle as well the canceled transactions, when re-routing makes no sense -- you should do it (see default config file for example. Cheers, Daniel We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. Any suggestion? -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [ mailto:sr-users-bounces@lists.sip-router.org ] On Behalf Of Alex Balashov Sent: Thursday, June 02, 2011 3:30 PM To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Problem using dispatcher. On 06/02/2011 03:26 PM, Gary Chen wrote: > the dispatcher always get into FAILURE route although the log shows > that kamailio did send INVITE to 10.10.1.222 and received '407 - > Proxy Authentication Required' from Freexwitch1 and then dispatcher > send INVITE to 10.10.1.223 and also received '407 - Proxy > Authentication Required' from Freexwitch2 and eventually failed. That's because any reply >= 300 is considered a failure. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

Thanks for the quick reply. How can I fix this? We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. Any suggestion? -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Alex Balashov Sent: Thursday, June 02, 2011 3:30 PM To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Problem using dispatcher. On 06/02/2011 03:26 PM, Gary Chen wrote: > the dispatcher always get into FAILURE route although the log shows > that kamailio did send INVITE to 10.10.1.222 and received '407 - > Proxy Authentication Required' from Freexwitch1 and then dispatcher > send INVITE to 10.10.1.223 and also received '407 - Proxy > Authentication Required' from Freexwitch2 and eventually failed. That's because any reply >= 300 is considered a failure. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

Kamailio 3.1.2 Here is my setup: Phone1 (10.10.1.114) | Kamailio (10.10.1.230) / \ / \ Freeswitch1 Freeswitch2 ----------------------> Other phones (10.10.1.222) (10.10.1.223) Both Freeswitch's share the same REGISTRATION information. Phone1 is already registered with Freeswitch. When I try to dial 'Other phones' using phone1, the dispatcher always get into FAILURE route although the log shows that kamailio did send INVITE to 10.10.1.222 and received '407 - Proxy Authentication Required' from Freexwitch1 and then dispatcher send INVITE to 10.10.1.223 and also received '407 - Proxy Authentication Required' from Freexwitch2 and eventually failed. Any idea? Here is my configuration: modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 10000) modparam("dispatcher", "ds_probing_threshhold", 3) #modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:proxy@SIP-router1.testnet.net") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "flags", 3) modparam("dispatcher", "force_dst", 1) route { .... if ( ds_select_dst("$var(group)", "9") ){ route(RELAY_DS); return; ... } route[[RELAY_DS] if (is_method("INVITE")) { t_on_branch("BRANCH_ONE"); t_on_reply("NAT_REPLY"); t_on_failure("FAIL_DS"); } if (!t_relay()) { sl_reply_error(); } } failure_route[FAIL_DS] { if (t_check_status("408")) { ds_mark_dst("i"); } if (!ds_next_dst()) { t_reply("503", "Service unavailable: no more dst"); exit; } t_on_failure("FAIL_DS"); route(RELAY_DS); } onreply_route[NAT_REPLY] { xlog("L_INFO", "REPLY_ROUTE: S=$rs - $rr F=$fu T=$tu IP=$avp(s:ip):$avp(s:port) ($si:$sp) ID=$ci\n"); } And here is the SIP messages from ngrep: U 10.10.1.144:51029 -> 10.10.1.230:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..F rom: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet .net;user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 70..Date: Thu, 02 Jun 2011 19:14:14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires: 180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces..C ontent-Length: 261..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.144..s=SIP Call..t=0 0..m=audio 20372 RTP/AVP 0 8 18 101..c=IN IP4 10.10.1.144..a=rtpmap: 0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv .. # U 10.10.1.230:5060 -> 10.10.1.144:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net; user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..Server: kamailio (3.1.2 (x8 6_64/linux))..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.222:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Record-Route: <sip:10.10.1.230;lr=on;nat=yes>..Via: SIP/2.0/ UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "L ine1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;us er=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..Date: Thu, 02 Jun 2011 19:14: 14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires: 180..A ccept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces..Content- Length: 277..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.231..s=SIP Call..t=0 0..m=audio 38426 RTP/AVP 0 8 18 101..c=IN IP4 10.10.1.231..a=rtpmap:0 PCMU/80 00..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..a=nortp proxy:yes.. # U 10.10.1.222:5060 -> 10.10.1.230:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/UDP 10.10.1.144:5060 ;branch=z9hG4bK116d554d..Record-Route: <sip:10.10.1.230;lr=on;nat=yes>..From: "Line1" <sip:5022441011@fs2000.lightyea r.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>..Call-ID: 00082166-efcb0013 -692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 0 3-05-32 -0400..Content-Length: 0.... # U 10.10.1.222:5060 -> 10.10.1.230:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/U DP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb0021 53bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>;tag=6mjQm2eayZ49j..Call-ID: 00082166-efcb0013-692f44d9 -6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 03-05-32 - 0400..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-se ize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest rea lm="fs2000.testnet.net", nonce="8da1ddb1-e393-442c-8be0-17e439457363", algorithm=MD5, qop="auth"..Content-Length: 0... . # U 10.10.1.230:5060 -> 10.10.1.222:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Fro m: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.n et;user=phone>;tag=6mjQm2eayZ49j..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..CSeq: 101 ACK..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.223:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Record-Route: <sip:10.10.1.230;lr=on;nat=yes;nat=yes>..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d.. From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.lightyea r.net;user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..Date: Thu, 02 Jun 201 1 19:14:14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires : 180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces.. Content-Length: 328..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.23110.10.1.231..s=SIP Call..t=0 0..m=audio 3842638426 RTP/AVP 0 8 18 101..c=IN IP4 216.49. 129.23110.10.1.231..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8 000..a=fmtp:101 0-15..a=sendrecv..a=nortpproxy:yes..a=nortpproxy:yes.. # U 10.10.1.223:5060 -> 10.10.1.230:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/UDP 10.10.1.144:5060 ;branch=z9hG4bK116d554d..Record-Route: <sip:10.10.1.230;lr=on;nat=yes;nat=yes>..From: "Line1" <sip:5022441011@fs2000. testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>..Call-ID: 00082166- efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011 -04-19 03-05-32 -0400..Content-Length: 0.... # U 10.10.1.223:5060 -> 10.10.1.230:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/U DP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb0021 53bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>;tag=Ky0XB4gDeDy6F..Call-ID: 00082166-efcb0013-692f44d9 -6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 03-05-32 - 0400..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-se ize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest rea lm="fs2000.testnet.net", nonce="8769298b-c52d-43b8-b69c-6ebf26f3184d", algorithm=MD5, qop="auth"..Content-Length: 0... . # U 10.10.1.230:5060 -> 10.10.1.223:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Fro m: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.n et;user=phone>;tag=Ky0XB4gDeDy6F..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..CSeq: 101 ACK..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.144:5060 SIP/2.0 503 Service unavailable: no more dst..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1 " <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=p hone>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..Server: kamailio (3.1.2 (x86_64/linux))..Content-Length: 0.... # U 10.10.1.144:51030 -> 10.10.1.230:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From : "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.ne t;user=phone>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..D ate: Thu, 02 Jun 2011 19:14:14 GMT..CSeq: 101 ACK..Content-Length: 0.... Gary Chen

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discussing about next major release - v3.2.0
by Daniel-Constantin Mierla 02 Jun '11

02 Jun '11

Hello, we should decide the roadmap to next major release v3.2.0, so probably an IRC meeting for some real time chatting would be good. I am thinking of June 3 or 4, 15:00GMT. Anyone interested in the meeting and available in one of those dates? The date with most of devs available will be selected. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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RTP traffic
by Lucas Alvarez 01 Jun '11

01 Jun '11

Hi, I'm having a problem with RTP traffic, I have softphone connecting from the outside, I can register but Kamailio is sending all RTP traffic to the private IP address that the softphone has in it own local network. How can I fix this? Any help will be appreciated. Lucas

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serweb & xSER 3.1
by Skyler 01 Jun '11

01 Jun '11

Hi all, Wondering if anyone has serweb working/functioning with Kamailio or SER 3.1 branch? If so which version of serweb are you using 2.0.0 or 2.1.0-dev1? Any issues or tips for first time install/setup of serweb? S.

1 0

has_tran_tmcbs function
by Bruno Bresciani 01 Jun '11

01 Jun '11

Hi, I can't understand the has_tran_tmcbs function... Can someone tell me how it works? Regards,

2 1

Kamailio Tables with index
by Klaus Darilion 01 Jun '11

01 Jun '11

Hi! With Openser 1.2 all the tables got an auto-increment-index column. Is this index column just for layout/performance purposes or is it also used somewhere insides the Kamailio code? If not (what I expect), then Kamailio 3.1 should still work with tables without those columns - correct? Thanks Klaus

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ClueCon 2011 News and Notes
by marketing＠cluecon.com 31 May '11

31 May '11

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Issue using pua_bla module
by Watkins, Bradley 31 May '11

31 May '11

I'm trying to get the pua_bla module to do its thing, and I'm running into an issue. I'm getting these errors when the first NOTIFY comes from the phone: ERROR: presence [publish.c:515]: Missing or unsupported event header field value ERROR: presence [publish.c:518]: event=[dialog;sla] I see the phone register and the subsequent SUBSCRIBE (both successful) from Kamailio, so this bit seems to be correct. I'm loading what I think are all the necessary modules: loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "pua.so" loadmodule "pua_bla.so" And I'm handling the NOTIFY as per the documentation (xlogs to see that in fact the script executes): if(is_method("NOTIFY") && to_uri=~"bla_aor(a)domain.com") { xlog("Matched bla_aor to_uri\n"); if(bla_handle_notify()) { xlog("Handled NOTIFY\n"); t_reply("200", "OK"); } } I see the log entries from the above xlogs, so it certainly is executing this part of the script. Any ideas why I'm getting the errors? All ideas and/or assistance are greatly appreciated. Thanks in advance, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

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Siremis: changes to Kamailio DB
by Klaus Darilion 31 May '11

31 May '11

Hi! I just read the Siremis installation howto. One of the task of the installer is: "Update SIP Database - this updates the structure of Kamailio database for accounting and charts modules" I wonder what siremis exactly does to the DB. Is greped over the siremis source file but couldn't find the changes. Are they somewhere documented? (e.g. .sql file with the statemens). Anyway, I wonder changes are needed at all. IMO, if possible, the Kamailio DB should not be changed by an external too - otherwise this may cause problems with updates to the Kamailio DB structure. Thanks Klaus

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kamailio debian package bug report : 3.1.3+lenny2
by Philippe HENSEL 31 May '11

31 May '11

Hi developpers and packagers, The file "/usr/lib64/kamailio/libsrdb2.so.1.0" seems to be included in both "kamailio_3.1.3+lenny2_amd64.deb" and "kamailio-mysql-modules_3.1.3 +lenny2_amd64.deb". So it is not possible to install "kamailio-mysql-modules". Thanks for help

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Multidomains - quick question
by Lucas Alvarez 31 May '11

31 May '11

Hi, if I want to register clients from outside of my local network, do I have to enable the domain module and add the external IP as a new domain? Thanks in advance, Lucas Alvarez

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acc Locks
by Borin 31 May '11

31 May '11

Hi all, Recently I met the following problem with kamailio acc table. We are using kamailio for calls dispatching to a couple of rtp gateways and for call statistics acc is used (with mysql). I tried to delete some old entries from acc table and for this time mysql "show processes" showed a lot of entries like " kamailio | Query | 0 | System lock | insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,start_time,src_addr,dst_addr,src_ |". I think that meant that kamailio could not write to the acc table for the time the data were being deleted. And it also did not forward INVITE requests further. Does that mean that if kamailio does not manage to write acc record for any reason it does not forward that INVITE/200ok? Best Regards, Kate

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Green VoIP - energy efficiency and performaces of v3.0
by Daniel-Constantin Mierla 31 May '11

31 May '11

Hello, Jan Janak conducted a very interesting research project regarding energy efficiency of VoIP systems during 2010, a collaboration between iptel.org and Columbia University. The team used the source code from sip-router.org GIT repository from January 2010, which corresponds to Kamailio (former OpenSER) and SER v3.0. The latest stable series v3.1 shares the same internal architecture with v3.0. As part of the research work, Jan could also gather some figures about capacity and performances of v3.0 with a quite complex configuration file: etc/sip-router-oob.cfg (involving authentication and NAT traversal as well). You can read the paper about energy efficiency at: - Green VoIP Article: http://asipto.com/u/2j The draft notes about capacity and performances of v3.0 are available at: - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k Some interesting results: - one instance of SIP server with 500 000 online users (mixed users – behind and not NAT routers) – consumed energy 210W - one instance of SIP server with 1 000 000 online users (no NAT involved) – consumed energy 190W - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for SIP server, the server could support 43 000 simultaneous TLS connections – consumed energy 203W - one SIP server instance with 80 000 permanent TCP connections, the SIP server could still handle at least 1000 requests per second and a connection arrival rate of 1000 new connections per second, done for 20 000 new connections. CPU load generated by the SIP server was from 6% to 8%. I added a new section to the draft notes to list the enhancements done for the latest stable release (v3.1.x) that contribute to performance improvements, like asynchronous TLS, fine tuning of memory for TLS connections and raw UDP sockets. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

5 9

Late parallel branch (without canceling previous pending branches)
by Iñaki Baz Castillo 31 May '11

31 May '11

Hi, just wondering whether the following scenario (which it's not possible in kamailio 1.5) is possible in 3.X: 1) Alice calls a PSTN number. 2) Proxy receives the call and routes it to a media server which plays an early-announcement (30 seconds long). 3) After N seconds (maybe 10 seconds) proxy starts a new parallel branch to the PSTN gateway *without* cancelling the previous branch with the media server. 4) Let's suppose the PSTN phone rings and proxy gets 183 with SDP (ringing in real audio). Proxy converts such 183 into a 180 with no SDP (already possible in Kamailio). 5) After 30 seconds media server rejects the call with 480. PSTN branch remains ringing as usual. I just would like to know if Kamailio 3.X allows point 3. I think it's hard but maybe I miss some feature. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

Dispatcher variables
by Lee Archer 31 May '11

31 May '11

Hi all, I am using dispatcher and on a failure I would like to log the server that has failed. I am using $dd but this only ever returns the address of the first server that was tried and not any subsequent servers that were tried and also failed. What should I be using to display the IP address of the server dispatcher is about to use? Thanks Lee

2 2

Fw: Load capacity of SER??
by Piyush.Bansal＠relianceada.com 31 May '11

31 May '11

Dear All, I'm using SER as a SIP proxy server for quite a long time. I'm not been able to find its load bearing capacity. If you can help me get a figure in terms of BHCA, that'll be quite helpful. Thanks and Regards, --Piyush Bansal The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

1 0

IMC module doesn't save data to mysql database
by Ozren Lapcevic 30 May '11

30 May '11

Hi all, I'm trying to configure IMC module and get it to work properly but I've run into problems with saving IMC module data to mysql database. I'm using following code provided in IMC module documentation: modparam("imc", "db_url", DBURL) ## route block ### if(is_method("MESSAGE") && (uri=~ "sip:chat-[0-9]+@") || (uri=~ "sip:chat-manager@")) { xlog("L_INFO", "going to imc: F [$fu] T [$tu] B [$rb]\n"); if(imc_manager()) sl_send_reply("200", "ok - imc_manager()"); else sl_send_reply("500", "command error"); exit; } I am creating chat room with following message sent to chat-manager(a)abc.hr: "#create chat-111" DEBUG log returns following: May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: INFO: <script>: going to imc: F [sip:oz@abc.hr] T [sip:chat-manager@abc.hr] B [#create chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc.c:525]: found command May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:146]: command: [create] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:151]: parameter 0=[chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:171]: new room [chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:186]: added room uri= sip:chat-111@abc.hr May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_mng.c:322]: [uri]= sip:oz@abc.hr May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_mng.c:326]: [user]= oz May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:196]: added the owner as the first member [ sip:oz@abc.hr] "./kamctl fifo imc_list_rooms" returns following: ROOM:: URI=sip:chat-111@abc.hr MEMBERS=0 OWNER=sip:oz@abc.hr "./kamctl fifo imc_list_members sip:chat-111@abc.hr" returns following: ROOM:: sip:chat-111@abc.hr NR_OF_MEMBERS=1 MEMBER:: sip:oz@abc.hr However, that data is not persisted/saved in the imc_rooms and imc_members tables in the database. Any clues what might be the problem here? Any help is greatly appreciated. Best regards, Ozren Lapcevic PS I'm runing Kamailio 3.1.4 with default cfg file with enabled mysql, multidomain, usrlocdb, auth and NAT. I can register clients, establish calls, send IMs, etc. imc_rooms and imc_members tables were created when kamailio was installed. I've also checked imc-create.sql script and tables are correctly created. The same DBURL used for imc is also used for auth_db, domain and usrloc modules and I haven't experienced any problems with those modules and database connection/persistence. domain, location, version and subscriber tables are correctly populated. I'm not getting any errors from imc_manager(), it returns 200 OK. I can't trace any errors in DEBUG related to problems with mysql database.

1 0

Kamailio service not starting on reboot when using mysql database
by Rick Aldred 30 May '11

30 May '11

I have set up kamailio 3.1.2 on an install of Debian Squeeze, using the default config but with the following defines enabled: WITH_MYSQLWITH_AUTHWITH_USRLOCDBWITH_NAT Everything works as expected when starting Kamailio from a terminal, e.g.service kamailio start But after a reboot, the service fails to start and all SIP requests are dropped with a ICMP port unreachable error. The following is an extract of /var/log/syslog when this problem occurs. It looks like Kamailio is failing to access the mysql socket, as the mysql service has not yet started? Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2)Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [db.c:289]: could not add connection to the poolApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: usrloc [dlist.c:486]: failed to open database connectionApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: registrar [reg_mod.c:396]: failed to register domainApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:569Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:572Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:574Apr 12 10:08:34 SIPServer mysqld_safe: Starting mysqld daemon with databases from /var/lib/mysqlApr 12 10:08:34 SIPServer mysqld: 110412 10:08:34 [Note] Plugin 'FEDERATED' is disabled.Apr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 InnoDB: Started; log sequence number 0 44233Apr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 [Note] Event Scheduler: Loaded 0 eventsApr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 [Note] /usr/sbin/mysqld: ready for connections.Apr 12 10:08:36 SIPServer mysqld: Version: '5.1.49-3' socket: '/var/run/mysqld/mysqld.sock' port: 3306 (Debian) Both the Kamailio and mysql services start with priority 18 on run levels 2345. I tried adjusting the Kamailio service to priority 19, but the problem still occurs. I am relatively new to Kamailio and mysql as well as Linux / Debian administration, so I'm not really sure how to solve this problem. Any ideas would be greatly appreciated. Regards,Rick Aldred

5 10

kamailio and radius
by Asun 30 May '11

30 May '11

Hello I've installed kamailio 3.1 but It's doesn't Work with radius authentication. I've followed this manual <http://www.kamailio.org/docs/openser-radius-1.0.x.html> http://www.kamailio.org/docs/openser-radius-1.0.x.html but it does'nt work, because is for openser not kamailio. Is there another manual to do this? Anyone have radius authentication with kamailio? Can anyone help me? Thank you Asun

3 8

Presence changes not seen for first time login
by Swetha Raj 30 May '11

30 May '11

Hi All, I have Kamailio3.1 set up by following the steps mentioned here - http://kb.asipto.com/kamailio:presence:k31-made-simple. When i login two users via Jitsi and add each other,one user is not able to see the other user's presence changes.Below are the steps to reproduce - 1.Login user1@sipdomain via jitsi 2.Login user2@sipdomain vis jitsi 3.user1 adds user2,user2 receives the request and accepts it. 4.user2 adds user1. 5.Now any presence changes from user1 is visible to user2 But any changes from user2 is not visible to user1. This issue is resolved if the users re-login.Any ideas what could be the issue ? Also how do i see the complete SIP headers in kamailio log,is their a way to enable it ? Thanks, Swetha

3 4

Invitation to connect on LinkedIn
by Anand Kumar via LinkedIn 28 May '11

28 May '11

LinkedIn ------------ Anand Kumar requested to add you as a connection on LinkedIn: ------------------------------------------ Luis, I'd like to add you to my professional network on LinkedIn. - Anand Accept invitation from Anand Kumar http://www.linkedin.com/e/-y8k3ym-go88enz1-1n/XkppUi8mK5TI9VJsZIpnfBovln4Su… View invitation from Anand Kumar http://www.linkedin.com/e/-y8k3ym-go88enz1-1n/XkppUi8mK5TI9VJsZIpnfBovln4Su… ------------------------------------------ DID YOU KNOW you can conduct a more credible and powerful reference check using LinkedIn? Enter the company name and years of employment or the prospective employee to find their colleagues that are also in your network. This provides you with a more balanced set of feedback to evaluate that new hire. http://www.linkedin.com/e/-y8k3ym-go88enz1-1n/rsr/inv-27/ -- (c) 2011, LinkedIn Corporation

1 0

core kamailio 3.1.2
by Bruno Bresciani 27 May '11

27 May '11

Hi ALL Why kamailio 3.1.2 generate core when function recvmsg return 0? Below is the trace of the core kamailio: May 27 10:29:14 sswpst00 /home2/local/kamailio/sbin/kamailio[31288]: INFO: <script>: CFG: Cseq INVITE ... May 27 10:29:14 sswpst00 /home2/local/kamailio/sbin/kamailio[31288]: DEBUG: tm [t_reply.c:1129]: ->>>>>>>>> T_code=100, new_code=488 May 27 10:29:14 sswpst00 /home2/local/kamailio/sbin/kamailio[31288]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb61304a0, callback type 64, id 0 entered *May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 11* May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: DEBUG: <core> [tcp_main.c:3551]: DBG: handle_ser_child: dead child 1, pid 31288 (shutting down?) May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x82561e0, 11, -1, 0x0) fd_no=18 called *May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31285]: ALERT: <core> [main.c:741]: child process 31288 exited by a signal 11* May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31285]: ALERT: <core> [main.c:744]: core was generated May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31285]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD *May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: INFO: <core> [main.c:807]: INFO: signal 15 received* May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: DEBUG: <core> [main.c:818]: Memory status (pkg): Regards,

1 0

ser_ctl / serweb
by Claudio Furrer 27 May '11

27 May '11

Hi all, Just want to ask if ser_ctl application will be merged into sr3.1 or future releases. By the moment I've found it here [1], [2] or [3]. Is it full compatible with sip-router v3.x? I know kamctl and kamdbctl would be suggested but these are for kamailio flavour. I don't find too much work based on SER flavour regarding to database/users administration and web provisioning. BTW, siremis only works with kamailio db structures, but for sip-router v3 ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if is it working well with the current new releases of the project. I would appreciate if someone can advice me if kamailio is the way to go, because have find that sip-router or ser flavours miss some docs and apps that kamailio already have. I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ [2] http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… [3] http://cvs.berlios.de/viewvc/ser/serctl/ [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ [5] http://developer.berlios.de/projects/serweb/ Thank you, Claudio

3 2

Howto restore a FIFO file?
by Spinov Evgeniy 27 May '11

27 May '11

Hello, everyone. I've accidentally started 2nd copy of Kamailio and killed it at once. However, fifo file was deleted by killed Kamailio, while the first one copy is running fine. Is there any way to restore the fifo file to the first kamailio, unless restarting it? Thank you.

3 2

[OT] IETF SIMPLE WG will destroy MSRP with the new draft-ietf-simple-msrp-sessmatch-11
by Iñaki Baz Castillo 27 May '11

27 May '11

Hi, for those interested in MSRP protocol (instant message sessions and file transfer for SIP) there are bad news: Even if there are already clients and servers (MSRP relays o IM conference servers) implementing the MSRP protocol (RFC 4975 and 4976) the SIMPLE WG will publish a new draft [*] that breaks these RFC's just to satisfy big vendors interested in deploying MSRP capable SBC/ALG boxes (so instead of solving NAT issues with MSRP relays as RFC 4976 states, they want it to be fixed in the router by doing ugly ALG, or in a SBC). This is terrible because all the MSRP devices should implement this new draft in order to interoperate (no backward compatibility at all). The draft also breaks the security defined in RFC 4975 (for example, TLS name based authentication cannot work anymore). For further information I recommend reading these two posts: http://blog.tekelec.com/blog/bid/29816/More-on-MSRP-Session-Match-Extension http://blog.tekelec.com/blog/bid/33138/MSRP-Session-Match-Backwards-Compati… and also these very *hot* mail threads in the SIMPLE maillist: http://www.ietf.org/mail-archive/web/simple/current/msg09227.html http://www.ietf.org/mail-archive/web/simple/current/msg09229.html [*] http://tools.ietf.org/html/draft-ietf-simple-msrp-sessmatch-11 -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 23

Re: [SR-Users] sr-users Digest, Vol 72, Issue 67
by Boris 27 May '11

27 May '11

sr-users-request(a)lists.sip-router.org wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Aliases Using the Default Script; (David J.) > 2. Re: Green VoIP - energy efficiency and performances of v3.0 > (Jan Janak) > 3. Kamailio v3.1.4 Released (Daniel-Constantin Mierla) > 4. Re: ser_ctl / serweb (caio) > 5. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (I?aki Baz Castillo) > 6. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 26 May 2011 14:27:55 -0400 >From: "David J." <david(a)styleflare.com> >Subject: [SR-Users] Aliases Using the Default Script; >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <4DDE9BAB.4060401(a)styleflare.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >I am using the kamailio default script on 3.1.3. > >I was wondering what happens when I added an Alias in dbaliases? > >For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com > >when an invite comes in; it works perfect I got a 200 back. (1001 Device >rings.) > >If I add another alias to a remote server; ie. > >18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com > >I get a 404 back. > >I wonder what happens internally? > >What should I modify to enable this case; > >Thanks. > > > >------------------------------ > >Message: 2 >Date: Thu, 26 May 2011 20:49:39 +0200 >From: Jan Janak <jan(a)ryngle.com> >Subject: Re: [SR-Users] Green VoIP - energy efficiency and > performances of v3.0 >To: miconda(a)gmail.com >Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org> >Message-ID: <BANLkTikUb_iwBDEuqk977-ECmmHo4dEXvQ(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >No, I turned it off. > >-Jan > >On Thu, May 26, 2011 at 11:50, Daniel-Constantin Mierla ><miconda(a)gmail.com> wrote: >> Hello, >> >> out of curiosity, since you used the sources from GIT - was memory debugging >> on? It is usually enabled in master branch and that could have some impact >> in memory usage and performances... >> >> Thanks, >> Daniel >> >> On 5/25/11 3:00 PM, Jan Janak wrote: >>> >>> On Wed, May 25, 2011 at 06:54, Jeremya<jeremy(a)electrosilk.net> ?wrote: >>>> >>>> These figures pale into insignificance compared to the power required >>>> for standard SIP devices - typically 5-8 watts per device multiplied by >>>> the number of devices. >>>> >>>> When you factor in Gigabit Ethernet the power ups significantly. >>>> >>>> Optimisation at the server level is not significant on any scale. >>>> Optimisation on communications power: i.e. end-devices, DSL& ?switches >>>> is where the power savings are important. >>> >>> Sure, the total power consumption of the whole system is dominated by >>> the power consumption of end-point devices, there's no doubt about >>> that and the paper says that. >>> >>> Nevertheless, as an ITSP you are typically paying for the energy >>> consumed by your servers and in that case knowing what you can expect >>> and how many servers you need is useful. Modern data-center servers >>> have significant base-line power consumption and a portion of that >>> needs to be attributed to the SIP service running on those servers. >>> >>> -Jan >>> >>>> Daniel-Constantin Mierla wrote: >>>>> >>>>> Hello, >>>>> >>>>> Jan Janak conducted a very interesting research project regarding >>>>> energy efficiency of VoIP systems during 2010, a collaboration between >>>>> iptel.org and Columbia University. >>>>> >>>>> The team used the source code from sip-router.org GIT repository from >>>>> January 2010, which corresponds to Kamailio (former OpenSER) and SER >>>>> v3.0. The latest stable series v3.1 shares the same internal >>>>> architecture with v3.0. >>>>> >>>>> As part of the research work, Jan could also gather some figures about >>>>> capacity and performances of v3.0 with a quite complex configuration >>>>> file: etc/sip-router-oob.cfg (involving authentication and NAT >>>>> traversal as well). >>>>> >>>>> You can read the paper about energy efficiency at: >>>>> >>>>> - Green VoIP Article: http://asipto.com/u/2j >>>>> >>>>> The draft notes about capacity and performances of v3.0 are available >>>>> at: >>>>> >>>>> - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k >>>>> >>>>> Some interesting results: >>>>> >>>>> - one instance of SIP server with 500 000 online users (mixed users ? >>>>> behind and not NAT routers) ? consumed energy 210W >>>>> - one instance of SIP server with 1 000 000 online users (no NAT >>>>> involved) ? consumed energy 190W >>>>> - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for >>>>> SIP server, the server could support 43 000 simultaneous TLS >>>>> connections ? consumed energy 203W >>>>> - one SIP server instance with 80 000 permanent TCP connections, the >>>>> SIP server could still handle at least 1000 requests per second and a >>>>> connection arrival rate of 1000 new connections per second, done for >>>>> 20 000 new connections. CPU load generated by the SIP server was from >>>>> 6% to 8%. >>>>> >>>>> I added a new section to the draft notes to list the enhancements done >>>>> for the latest stable release (v3.1.x) that contribute to performance >>>>> improvements, like asynchronous TLS, fine tuning of memory for TLS >>>>> connections and raw UDP sockets. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > > > >------------------------------ > >Message: 3 >Date: Thu, 26 May 2011 21:00:42 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >Subject: [SR-Users] Kamailio v3.1.4 Released >To: kamailio <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org >Message-ID: <4DDEA35A.7070606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >Kamailio SIP Server v3.1.4 stable release is out. > >This is a maintenance release of latest stable branch, 3.1, that >includes fixes since release of v3.1.3. There is no change to database >schema or configuration language structure. Deployments running previous >v3.1.x versions are strongly recommended to be upgraded to v3.1.4. > >For more details about version 3.1.4, visit: > >http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ > >Cheers, >Daniel > >-- >Daniel-Constantin Mierla -- http://www.asipto.com >http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >------------------------------ > >Message: 4 >Date: Thu, 26 May 2011 17:59:42 -0300 >From: caio <elcaio(a)gmail.com> >Subject: Re: [SR-Users] ser_ctl / serweb >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTik+dnTp-egK2xy1ua=tPnCcaDuuOA(a)mail.gmail.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Anyone know if ser_ctl (python utility) will be merged into master? Or is >going to be deprecated? >If it still alive, are there any examples/docs of the usage for DB users >provisoining? >kamctl and kamdbctl are recomended for its substitution? > >Which utility will be official for sip-router? >Sorry if it already were discussed, but I didn't find comments about it >since 2010. > >Thank you >Claudio > >On Tue, May 24, 2011 at 11:14 AM, Claudio Furrer <elcaio(a)gmail.com> wrote: > >> Hi all, >> >> Just want to ask if ser_ctl application will be merged into sr3.1 or future >> releases. >> By the moment I've found it here [1], [2] or [3]. >> >> Is it full compatible with sip-router v3.x? >> >> I know kamctl and kamdbctl would be suggested but these are for kamailio >> flavour. I don't find too much work based on SER flavour regarding to >> database/users administration and web provisioning. >> BTW, siremis only works with kamailio db structures, but for sip-router v3 >> ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if >> is >> it working well with the current new releases of the project. >> >> I would appreciate if someone can advice me if kamailio is the way to go, >> because have find that sip-router or ser flavours miss some docs and apps >> that >> kamailio already have. >> >> I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( >> >> [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ >> [2] >> >> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… >> [3] http://cvs.berlios.de/viewvc/ser/serctl/ >> >> [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ >> [5] http://developer.berlios.de/projects/serweb/ >> >> Thank you, >> Claudio >> >

1 0

Re: [SR-Users] sr-users Digest, Vol 72, Issue 67
by Boris 27 May '11

27 May '11

sr-users-request(a)lists.sip-router.org wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Aliases Using the Default Script; (David J.) > 2. Re: Green VoIP - energy efficiency and performances of v3.0 > (Jan Janak) > 3. Kamailio v3.1.4 Released (Daniel-Constantin Mierla) > 4. Re: ser_ctl / serweb (caio) > 5. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (I?aki Baz Castillo) > 6. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 26 May 2011 14:27:55 -0400 >From: "David J." <david(a)styleflare.com> >Subject: [SR-Users] Aliases Using the Default Script; >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <4DDE9BAB.4060401(a)styleflare.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >I am using the kamailio default script on 3.1.3. > >I was wondering what happens when I added an Alias in dbaliases? > >For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com > >when an invite comes in; it works perfect I got a 200 back. (1001 Device >rings.) > >If I add another alias to a remote server; ie. > >18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com > >I get a 404 back. > >I wonder what happens internally? > >What should I modify to enable this case; > >Thanks. > > > >------------------------------ > >Message: 2 >Date: Thu, 26 May 2011 20:49:39 +0200 >From: Jan Janak <jan(a)ryngle.com> >Subject: Re: [SR-Users] Green VoIP - energy efficiency and > performances of v3.0 >To: miconda(a)gmail.com >Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org> >Message-ID: <BANLkTikUb_iwBDEuqk977-ECmmHo4dEXvQ(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >No, I turned it off. > >-Jan > >On Thu, May 26, 2011 at 11:50, Daniel-Constantin Mierla ><miconda(a)gmail.com> wrote: >> Hello, >> >> out of curiosity, since you used the sources from GIT - was memory debugging >> on? It is usually enabled in master branch and that could have some impact >> in memory usage and performances... >> >> Thanks, >> Daniel >> >> On 5/25/11 3:00 PM, Jan Janak wrote: >>> >>> On Wed, May 25, 2011 at 06:54, Jeremya<jeremy(a)electrosilk.net> ?wrote: >>>> >>>> These figures pale into insignificance compared to the power required >>>> for standard SIP devices - typically 5-8 watts per device multiplied by >>>> the number of devices. >>>> >>>> When you factor in Gigabit Ethernet the power ups significantly. >>>> >>>> Optimisation at the server level is not significant on any scale. >>>> Optimisation on communications power: i.e. end-devices, DSL& ?switches >>>> is where the power savings are important. >>> >>> Sure, the total power consumption of the whole system is dominated by >>> the power consumption of end-point devices, there's no doubt about >>> that and the paper says that. >>> >>> Nevertheless, as an ITSP you are typically paying for the energy >>> consumed by your servers and in that case knowing what you can expect >>> and how many servers you need is useful. Modern data-center servers >>> have significant base-line power consumption and a portion of that >>> needs to be attributed to the SIP service running on those servers. >>> >>> -Jan >>> >>>> Daniel-Constantin Mierla wrote: >>>>> >>>>> Hello, >>>>> >>>>> Jan Janak conducted a very interesting research project regarding >>>>> energy efficiency of VoIP systems during 2010, a collaboration between >>>>> iptel.org and Columbia University. >>>>> >>>>> The team used the source code from sip-router.org GIT repository from >>>>> January 2010, which corresponds to Kamailio (former OpenSER) and SER >>>>> v3.0. The latest stable series v3.1 shares the same internal >>>>> architecture with v3.0. >>>>> >>>>> As part of the research work, Jan could also gather some figures about >>>>> capacity and performances of v3.0 with a quite complex configuration >>>>> file: etc/sip-router-oob.cfg (involving authentication and NAT >>>>> traversal as well). >>>>> >>>>> You can read the paper about energy efficiency at: >>>>> >>>>> - Green VoIP Article: http://asipto.com/u/2j >>>>> >>>>> The draft notes about capacity and performances of v3.0 are available >>>>> at: >>>>> >>>>> - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k >>>>> >>>>> Some interesting results: >>>>> >>>>> - one instance of SIP server with 500 000 online users (mixed users ? >>>>> behind and not NAT routers) ? consumed energy 210W >>>>> - one instance of SIP server with 1 000 000 online users (no NAT >>>>> involved) ? consumed energy 190W >>>>> - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for >>>>> SIP server, the server could support 43 000 simultaneous TLS >>>>> connections ? consumed energy 203W >>>>> - one SIP server instance with 80 000 permanent TCP connections, the >>>>> SIP server could still handle at least 1000 requests per second and a >>>>> connection arrival rate of 1000 new connections per second, done for >>>>> 20 000 new connections. CPU load generated by the SIP server was from >>>>> 6% to 8%. >>>>> >>>>> I added a new section to the draft notes to list the enhancements done >>>>> for the latest stable release (v3.1.x) that contribute to performance >>>>> improvements, like asynchronous TLS, fine tuning of memory for TLS >>>>> connections and raw UDP sockets. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > > > >------------------------------ > >Message: 3 >Date: Thu, 26 May 2011 21:00:42 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >Subject: [SR-Users] Kamailio v3.1.4 Released >To: kamailio <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org >Message-ID: <4DDEA35A.7070606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >Kamailio SIP Server v3.1.4 stable release is out. > >This is a maintenance release of latest stable branch, 3.1, that >includes fixes since release of v3.1.3. There is no change to database >schema or configuration language structure. Deployments running previous >v3.1.x versions are strongly recommended to be upgraded to v3.1.4. > >For more details about version 3.1.4, visit: > >http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ > >Cheers, >Daniel > >-- >Daniel-Constantin Mierla -- http://www.asipto.com >http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >------------------------------ > >Message: 4 >Date: Thu, 26 May 2011 17:59:42 -0300 >From: caio <elcaio(a)gmail.com> >Subject: Re: [SR-Users] ser_ctl / serweb >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTik+dnTp-egK2xy1ua=tPnCcaDuuOA(a)mail.gmail.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Anyone know if ser_ctl (python utility) will be merged into master? Or is >going to be deprecated? >If it still alive, are there any examples/docs of the usage for DB users >provisoining? >kamctl and kamdbctl are recomended for its substitution? > >Which utility will be official for sip-router? >Sorry if it already were discussed, but I didn't find comments about it >since 2010. > >Thank you >Claudio > >On Tue, May 24, 2011 at 11:14 AM, Claudio Furrer <elcaio(a)gmail.com> wrote: > >> Hi all, >> >> Just want to ask if ser_ctl application will be merged into sr3.1 or future >> releases. >> By the moment I've found it here [1], [2] or [3]. >> >> Is it full compatible with sip-router v3.x? >> >> I know kamctl and kamdbctl would be suggested but these are for kamailio >> flavour. I don't find too much work based on SER flavour regarding to >> database/users administration and web provisioning. >> BTW, siremis only works with kamailio db structures, but for sip-router v3 >> ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if >> is >> it working well with the current new releases of the project. >> >> I would appreciate if someone can advice me if kamailio is the way to go, >> because have find that sip-router or ser flavours miss some docs and apps >> that >> kamailio already have. >> >> I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( >> >> [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ >> [2] >> >> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… >> [3] http://cvs.berlios.de/viewvc/ser/serctl/ >> >> [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ >> [5] http://developer.berlios.de/projects/serweb/ >> >> Thank you, >> Claudio >> >

1 0

Re: [SR-Users] sr-users Digest, Vol 72, Issue 67
by Boris 27 May '11

27 May '11

sr-users-request(a)lists.sip-router.org wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Aliases Using the Default Script; (David J.) > 2. Re: Green VoIP - energy efficiency and performances of v3.0 > (Jan Janak) > 3. Kamailio v3.1.4 Released (Daniel-Constantin Mierla) > 4. Re: ser_ctl / serweb (caio) > 5. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (I?aki Baz Castillo) > 6. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 26 May 2011 14:27:55 -0400 >From: "David J." <david(a)styleflare.com> >Subject: [SR-Users] Aliases Using the Default Script; >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <4DDE9BAB.4060401(a)styleflare.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >I am using the kamailio default script on 3.1.3. > >I was wondering what happens when I added an Alias in dbaliases? > >For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com > >when an invite comes in; it works perfect I got a 200 back. (1001 Device >rings.) > >If I add another alias to a remote server; ie. > >18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com > >I get a 404 back. > >I wonder what happens internally? > >What should I modify to enable this case; > >Thanks. > > > >------------------------------ > >Message: 2 >Date: Thu, 26 May 2011 20:49:39 +0200 >From: Jan Janak <jan(a)ryngle.com> >Subject: Re: [SR-Users] Green VoIP - energy efficiency and > performances of v3.0 >To: miconda(a)gmail.com >Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org> >Message-ID: <BANLkTikUb_iwBDEuqk977-ECmmHo4dEXvQ(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >No, I turned it off. > >-Jan > >On Thu, May 26, 2011 at 11:50, Daniel-Constantin Mierla ><miconda(a)gmail.com> wrote: >> Hello, >> >> out of curiosity, since you used the sources from GIT - was memory debugging >> on? It is usually enabled in master branch and that could have some impact >> in memory usage and performances... >> >> Thanks, >> Daniel >> >> On 5/25/11 3:00 PM, Jan Janak wrote: >>> >>> On Wed, May 25, 2011 at 06:54, Jeremya<jeremy(a)electrosilk.net> ?wrote: >>>> >>>> These figures pale into insignificance compared to the power required >>>> for standard SIP devices - typically 5-8 watts per device multiplied by >>>> the number of devices. >>>> >>>> When you factor in Gigabit Ethernet the power ups significantly. >>>> >>>> Optimisation at the server level is not significant on any scale. >>>> Optimisation on communications power: i.e. end-devices, DSL& ?switches >>>> is where the power savings are important. >>> >>> Sure, the total power consumption of the whole system is dominated by >>> the power consumption of end-point devices, there's no doubt about >>> that and the paper says that. >>> >>> Nevertheless, as an ITSP you are typically paying for the energy >>> consumed by your servers and in that case knowing what you can expect >>> and how many servers you need is useful. Modern data-center servers >>> have significant base-line power consumption and a portion of that >>> needs to be attributed to the SIP service running on those servers. >>> >>> -Jan >>> >>>> Daniel-Constantin Mierla wrote: >>>>> >>>>> Hello, >>>>> >>>>> Jan Janak conducted a very interesting research project regarding >>>>> energy efficiency of VoIP systems during 2010, a collaboration between >>>>> iptel.org and Columbia University. >>>>> >>>>> The team used the source code from sip-router.org GIT repository from >>>>> January 2010, which corresponds to Kamailio (former OpenSER) and SER >>>>> v3.0. The latest stable series v3.1 shares the same internal >>>>> architecture with v3.0. >>>>> >>>>> As part of the research work, Jan could also gather some figures about >>>>> capacity and performances of v3.0 with a quite complex configuration >>>>> file: etc/sip-router-oob.cfg (involving authentication and NAT >>>>> traversal as well). >>>>> >>>>> You can read the paper about energy efficiency at: >>>>> >>>>> - Green VoIP Article: http://asipto.com/u/2j >>>>> >>>>> The draft notes about capacity and performances of v3.0 are available >>>>> at: >>>>> >>>>> - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k >>>>> >>>>> Some interesting results: >>>>> >>>>> - one instance of SIP server with 500 000 online users (mixed users ? >>>>> behind and not NAT routers) ? consumed energy 210W >>>>> - one instance of SIP server with 1 000 000 online users (no NAT >>>>> involved) ? consumed energy 190W >>>>> - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for >>>>> SIP server, the server could support 43 000 simultaneous TLS >>>>> connections ? consumed energy 203W >>>>> - one SIP server instance with 80 000 permanent TCP connections, the >>>>> SIP server could still handle at least 1000 requests per second and a >>>>> connection arrival rate of 1000 new connections per second, done for >>>>> 20 000 new connections. CPU load generated by the SIP server was from >>>>> 6% to 8%. >>>>> >>>>> I added a new section to the draft notes to list the enhancements done >>>>> for the latest stable release (v3.1.x) that contribute to performance >>>>> improvements, like asynchronous TLS, fine tuning of memory for TLS >>>>> connections and raw UDP sockets. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > > > >------------------------------ > >Message: 3 >Date: Thu, 26 May 2011 21:00:42 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >Subject: [SR-Users] Kamailio v3.1.4 Released >To: kamailio <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org >Message-ID: <4DDEA35A.7070606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >Kamailio SIP Server v3.1.4 stable release is out. > >This is a maintenance release of latest stable branch, 3.1, that >includes fixes since release of v3.1.3. There is no change to database >schema or configuration language structure. Deployments running previous >v3.1.x versions are strongly recommended to be upgraded to v3.1.4. > >For more details about version 3.1.4, visit: > >http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ > >Cheers, >Daniel > >-- >Daniel-Constantin Mierla -- http://www.asipto.com >http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >------------------------------ > >Message: 4 >Date: Thu, 26 May 2011 17:59:42 -0300 >From: caio <elcaio(a)gmail.com> >Subject: Re: [SR-Users] ser_ctl / serweb >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTik+dnTp-egK2xy1ua=tPnCcaDuuOA(a)mail.gmail.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Anyone know if ser_ctl (python utility) will be merged into master? Or is >going to be deprecated? >If it still alive, are there any examples/docs of the usage for DB users >provisoining? >kamctl and kamdbctl are recomended for its substitution? > >Which utility will be official for sip-router? >Sorry if it already were discussed, but I didn't find comments about it >since 2010. > >Thank you >Claudio > >On Tue, May 24, 2011 at 11:14 AM, Claudio Furrer <elcaio(a)gmail.com> wrote: > >> Hi all, >> >> Just want to ask if ser_ctl application will be merged into sr3.1 or future >> releases. >> By the moment I've found it here [1], [2] or [3]. >> >> Is it full compatible with sip-router v3.x? >> >> I know kamctl and kamdbctl would be suggested but these are for kamailio >> flavour. I don't find too much work based on SER flavour regarding to >> database/users administration and web provisioning. >> BTW, siremis only works with kamailio db structures, but for sip-router v3 >> ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if >> is >> it working well with the current new releases of the project. >> >> I would appreciate if someone can advice me if kamailio is the way to go, >> because have find that sip-router or ser flavours miss some docs and apps >> that >> kamailio already have. >> >> I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( >> >> [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ >> [2] >> >> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… >> [3] http://cvs.berlios.de/viewvc/ser/serctl/ >> >> [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ >> [5] http://developer.berlios.de/projects/serweb/ >> >> Thank you, >> Claudio >> >

1 0

Kamailio v3.1.4 Released
by Daniel-Constantin Mierla 26 May '11

26 May '11

Hello, Kamailio SIP Server v3.1.4 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.3. There is no change to database schema or configuration language structure. Deployments running previous v3.1.x versions are strongly recommended to be upgraded to v3.1.4. For more details about version 3.1.4, visit: http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Aliases Using the Default Script;
by David J. 26 May '11

26 May '11

I am using the kamailio default script on 3.1.3. I was wondering what happens when I added an Alias in dbaliases? For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com when an invite comes in; it works perfect I got a 200 back. (1001 Device rings.) If I add another alias to a remote server; ie. 18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com I get a 404 back. I wonder what happens internally? What should I modify to enable this case; Thanks.

1 0

preparing v3.1.4
by Daniel-Constantin Mierla 26 May '11

26 May '11

Later today I will package v3.1.4, so if you have something to commit in branch 3.1, please do it before 15:00GMT. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

packaging 3.1.4
by Juha Heinanen 26 May '11

26 May '11

Daniel-Constantin Mierla writes: > I am thinking of packaging v3.1.4 next week on Thursday -- last minor > release was more than 1.5month ago. > Any comments against it? the bug still exists, where AVP_CLASS_DOMAIN avps are not saved in the transaction. i guess jan forgot about it. -- juha

2 1

"ACK not received"
by Mokhtar Bengana 26 May '11

26 May '11

I am trying to forward a DID to a sip account registered to Kamailio. The call get established with two way audio and Kamailio shows it's comming from an Asterisk server. However the call get disconnected each time after 31s. Ngrep is showing "ACK not received" by the Kamailio test account before the call get disconnected. If Kamailio is suppose to send an ACK back to the asterisk server to keep the call up, I am not sure why it's not happening. Or is it the other way around? I have included the ngrep output. I appreciate any help. U 2011/05/26 00:27:12.525344 91.121.81.212:5060 -> 192.168.2.4:5060 INVITE sip:test@sipsoft.dyndns.org SIP/2.0. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. From: "3313144859493" <sip:3313144859493@188.165.98.47>;tag=as6c63c119. To: <sip:33980307151@sip.antisip.com:5060>. Contact: <sip:3313144859493@188.165.98.47:5060>. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 69. Date: Thu, 26 May 2011 00:27:10 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Type: application/sdp. Content-Length: 260. P-Antisip: forwarded with alias. P-PMR: mediarelay enabled. . v=0. o=root 10502 10502 IN IP4 91.121.81.212. s=session. c=IN IP4 91.121.81.212. t=0 0. m=audio 41316 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2011/05/26 00:27:12.526734 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. From: "3313144859493" <sip:3313144859493@188.165.98.47>;tag=as6c63c119. To: <sip:33980307151@sip.antisip.com:5060>. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Server: kamailio (3.1.3 (x86_64/linux)). Content-Length: 0. . U 2011/05/26 00:27:12.527013 192.168.2.4:5060 -> 192.168.2.8:65412 INVITE sip:test@192.168.2.8:65412;rinstance=befda46036a026ac SIP/2.0. Record-Route: <sip:192.168.2.4;lr=on>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. From: "3313144859493" <sip:3313144859493@188.165.98.47>;tag=as6c63c119. To: <sip:33980307151@sip.antisip.com:5060>. Contact: <sip:3313144859493@188.165.98.47:5060>. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 68. Date: Thu, 26 May 2011 00:27:10 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Type: application/sdp. Content-Length: 260. P-Antisip: forwarded with alias. P-PMR: mediarelay enabled. . v=0. o=root 10502 10502 IN IP4 91.121.81.212. s=session. c=IN IP4 91.121.81.212. t=0 0. m=audio 41316 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2011/05/26 00:27:12.601815 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 0. . U 2011/05/26 00:27:12.601967 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 0. . U 2011/05/26 00:27:22.568944 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:22.569043 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:23.080398 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:23.080549 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:24.080628 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:24.080697 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:26.080654 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:26.080816 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:30.080527 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:30.080594 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:34.080318 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:34.080436 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:37.137775 192.168.2.8:65412 -> 192.168.2.4:5060 . . U 2011/05/26 00:27:38.081552 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:38.081707 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:42.082163 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:42.082308 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:46.080414 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:46.080562 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:50.082011 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:50.082142 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:54.080488 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:54.080592 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:54.574347 192.168.2.8:65412 -> 192.168.2.4:5060 BYE sip:3313144859493@188.165.98.47:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport. Max-Forwards: 70. Route: <sip:192.168.2.4;lr>. Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Reason: SIP;description="ACK not received". Content-Length: 0. . U 2011/05/26 00:27:54.574553 192.168.2.4:5060 -> 91.121.81.212:5060 BYE sip:3313144859493@188.165.98.47:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK79bb.32809f01.0. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport=65412. Max-Forwards: 69. Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Reason: SIP;description="ACK not received". Content-Length: 0. . U 2011/05/26 00:27:54.705228 91.121.81.212:5060 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;rport=5060;received=99.42.204.9;branch=z9hG4bK79bb.32809f01.0. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport=65412. Record-Route: <sip:91.121.81.212;lr>. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. . U 2011/05/26 00:27:54.705377 192.168.2.4:5060 -> 192.168.2.8:65412 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport=65412. Record-Route: <sip:91.121.81.212;lr>. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. . U 2011/05/26 00:28:07.139142 192.168.2.8:65412 -> 192.168.2.4:5060 . .

3 3

Wildcard TLS
by Jon Farmer 25 May '11

25 May '11

Hi I am just wondering if it is possible to use a wildcard certificate with TLS. I have hundreds of customers who are using subdomains of the main domain name on a virtual PBX platform. i.e if the main domain is example.org I have customers using cust1.example.org cust2.example.org cust3.example.org ....... I want to start offering TLS on a per customer basis. However I don't want to have to create a TLS certificate per customer if I don't need to. So I need to know if I can create a *.example.org certificate and use that for all customers who want TLS? Is this possible? Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 4

3.1.x and Kamailio compatibility mode
by Klaus Darilion 25 May '11

25 May '11

Hi! Is #!KAMAILIO still necessary with Kamailio 3.1.x? If yes, which behavior is changed by #!KAMAILIO? Thanks Klaus

3 8

Service Execution Time
by Giuseppe Carella 25 May '11

25 May '11

Hi all, I'm using kamailio with the dispatcher module. I would like to know if there are some API to get an average of the service execution time. In other words, I need to get the time between sending the request and receiving the response for every request, but also to have this values separate per server. I know also that this could be a little heavy for the dispatcher, but for some test I need this information. Your help is kindly appreciated! Thanks in advance, Giuseppe

2 1

Manipulate From header
by alex pappas 25 May '11

25 May '11

Dear all, I'm trying to change the From uri and Dsplay but without sucess. My config is as follow: if(dp_translate("1111", "$avp(s:frm_user_name)/$avp(s:test_frm_user_name)")) --> i'm sending 2112202701 and I get back corectly 701 { $avp(s:display) = $avp(s:test_frm_user_name); $avp(s:fu_uri) = "sip:" + $avp(s:test_frm_user_name) + "@" + $fd; xlog("alx ------- The avp(s:display): $avp(s:display) ---- avp(s:fu_uri)=$avp(s:fu_uri) ----------\n"); --> I see values 701 and sip:701@my_IP_address uac_replace_from("$avp(s:display)","$avp(s:fu_uri)"); Here I log the fu and I see that nothing has changed.. } Do you have any ideas on that ? I use Kamailio 3.1.3 Thank you in advance Alex

2 2

undefined symbol error
by Bruno Bresciani 24 May '11

24 May '11

Hi, Someone know tell me why kamailio generate the following error when I try load the route.so module: May 24 10:21:43 sswpst00 kamailio: ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </home2/local/kamailio/lib/kamailio/modules/route.so>: /home2/local/kamailio/lib/kamailio/modules/route.so: undefined symbol: db_use_table I don't know why this error have been generate. I'm porting this module to use with kamailio 3.1.2.

2 3

htable ::size has wrong value
by Juha Heinanen 24 May '11

24 May '11

i have entries in htable with key name lpi::20033::prefix: # sip-proxy_ctl mi sht_dump htable | egrep 20033 lpi::20033::prefix[0]:: +3583796294 lpi::20033::prefix[1]:: +35837824030 lpi::20033::prefix::size:: 1 as you see in above, size of that key name is 1 although there clearly is two entries in the table with that key name. according to readme ::size is number of items: 3.7. array_size_suffix (str) The suffix to be added to store the number of items in an array. i verified with another key name where there was only one prefix, and then ::size was again 1. so sometimes ::size is correct and sometimes incorrect. this looks like a serious bug to me. any comments? -- juha

2 4

Telefonia ip
by rom ignacio 24 May '11

24 May '11

friend I am interested in learning more about VoIP or IP telephony, I read a lot about you and I still have more to read, as it can not released through theofficial documentation page errors Please let me send them am email, thanks I DEVELOP voip server to sip in Venezuela, I need all the help possible. thanks *Rommel Malave .M ** * *T.S.U Universitario Extracción y Producción de Gas * * T.S.U en Informatica*** 04148241688

1 0

Re: [SR-Users] app_lua in kamailio
by Daniel-Constantin Mierla 24 May '11

24 May '11

Hello, please do not write direct private emails for questions about kamailio, use public mailing list sr-users(a)lists.sip-router.org (cc-ed) -- private emails are ignored, after the first notification. exit is not a function exported by a module, it is implemented in core. On the other hand, you try to stop the kamailio cfg interpreter when lua interpreter is executing the Lua script, so it won't work straightforward. Set a variable in kamailio from Lua, then check it after executing the lua script to decide to further exit or not. Cheers, Daniel On 5/24/11 11:44 AM, Bret McDanel wrote: > I am having a terrible time trying to get a conditional response from a > lua script. > > Basically I want > > if error then > sr.sl.send_reply(404,"Not Found"); > sr.modf("exit"); -- this does not work! > end > > > How can I have such a conditional? I have tried with return values from > the script to see if that would change anything but that did not seem to > work, sr.modf("exit") is not found. > > Any assistance would be greatly appreciated. > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 3

struct run_act_ctx
by Bruno Bresciani 24 May '11

24 May '11

Hi, I'd like to know the functionality of the struct run_act_ctx. I need to adapt my route module to use this struct, the function do_action require this parameter. I'm migrating of the kamailio 1.5.0 to 3.1.2. Regards,

2 2

Re: [SR-Users] 403 Proxy forbidden replay for invite that send sip gateway
by hala alramli 24 May '11

24 May '11

thanks for your reply Klaus ,attached my config file could you please take a look ,i found the line that send that replay but i do not know why .? could you help me i am stuck --- Mer 18/5/11, Klaus Darilion <klaus.mailinglists(a)pernau.at> ha scritto: Da: Klaus Darilion <klaus.mailinglists(a)pernau.at> Oggetto: Re: [SR-Users] 403 Proxy forbidden replay for invite that send sip gateway A: sr-users(a)lists.sip-router.org Data: Mercoledì 18 maggio 2011, 13:04 Check your config and verify where 403 is sent. You have something like sl_send_reply("403","Proxy forbidden") in your config. regards Klaus Am 18.05.2011 10:48, schrieb hala alramli: > Hello every body, > could you help me ? > > i bought Geographical DID numbers from gateway provider , to receive > calls from pstn > > my problem that when gateway send invite my proxy send 403 forbidden ,i > dont know why what is wrong ,any help please. > > this is a track for th gateway invite. > > my proxy ip:79.12.254.153 > gateway IP :77.239.128.10 > my geographical number :0698962030 > > > > > > INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: <sip:0698962030@10.39.6.207:5060;user=phone> > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 INVITE > > User-Agent: CS2000_NGSS/8.0 > > Remote-Party-ID: <sip:0289089500@10.39.1.16;user=phone>; party=calling; > privacy=off; screen=yes > > Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK > > P-Asserted-Identity: <sip:0289089500@10.39.1.16;user=phone> > > Max-Forwards: 69 > > Contact: <sip:77.239.128.10:5060;transport=UDP> > > Supported: 100rel > > Content-Type: application/sdp > > Content-Length: 236 > > > > v=0 > > o=PVG 1305215180830 1305215180830 IN IP4 77.239.128.10 > > s=- > > p=+1 6135555555 > > c=IN IP4 77.239.128.10 > > t=0 0 > > m=audio 49264 RTP/AVP 18 8 0 13 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > a=fmtp:18 annexb=no > > > > > > > > SIP/2.0 403 Proxy forbidden > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 INVITE > > Server: Kamailio (1.4.4-notls (x86_64/linux)) > > Content-Length: 0 > > > > > > > > > > > > > > > > > > > > > > ACK sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 ACK > > Max-Forwards: 69 > > Content-Length: 0 > > > > > > > INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: <sip:0698962030@10.39.6.207:5060;user=phone> > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 INVITE > > User-Agent: CS2000_NGSS/8.0 > > Remote-Party-ID: <sip:026683011@10.39.1.16;user=phone>; party=calling; > privacy=off; screen=yes > > Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK > > P-Asserted-Identity: <sip:026683011@10.39.1.16;user=phone> > > Max-Forwards: 69 > > Contact: <sip:77.239.128.10:5060;transport=UDP> > > Supported: 100rel > > Content-Type: application/sdp > > Content-Length: 236 > > > > v=0 > > o=PVG 1305215458700 1305215458700 IN IP4 77.239.128.10 > > s=- > > p=+1 6135555555 > > c=IN IP4 77.239.128.10 > > t=0 0 > > m=audio 41892 RTP/AVP 18 8 0 13 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > a=fmtp:18 annexb=no > > > > > > > > SIP/2.0 403 Proxy forbidden > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 INVITE > > Server: Kamailio (1.4.4-notls (x86_64/linux)) > > Content-Length: 0 > > > > > > > > > > > > > > > > > > > > > > > > sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 ACK > > Max-Forwards: 69 > > Content-Length: 0 > > > > > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 4

SEMS 1.4.0 release
by Stefan Sayer 24 May '11

24 May '11

We are happy to announce the availability of the SIP Express Media Server version 1.4.0. SEMS is a high performance media and application server for SIP based VoIP networks. This release features a powerful Session Border Controller (SBC) module. From completely transparent B2BUA to customized URI/From/To, strictly filtered (messages, headers, codecs) with RTP anchoring, Session Timer enforcement, prepaid and call timer, the SBC facilitates interconnect and core routing in a simple and secure way. Thanks to the new multihoming support, SEMS can now be employed at the border of the networks. This addition also allows to overcome the bottleneck of one NIC - giving the possibility to fully exploit SEMS' great performance. In the app development area, the DSM language has matured to become a viable candidate also for implementing complex application logic, thanks to language constructs like for, if and functions. SEMS can be downloaded in source from its ftp site at: ftp.iptel.org/pub/sems/1.4/1.4.0/ Packages for debian/ubuntu should appear shortly on the OBS page: https://build.opensuse.org/project/show?project=home:team-sems Contributors to this release: Stefan Sayer, Raphael Coeffic, Bogdan Pintea, Matthew Williams, Peter Lemenkov, Anton Zagorskiy, Robert Szokovacs, Jon Bonilla, Thomas Gelf, Erik Dekkers, Tobias Lindgren, David J Many thanks to the companies sponsoring SEMS development, especially FRAFOS GmbH and TelTech Systems Inc. "E stands for Express." * Apologies if you receive multiple copies of this message. *

4 5

~ operator has no effect
by Juha Heinanen 24 May '11

24 May '11

according to core cookbook: ~ : bitwise NOT however, looks like it has no effect. i have $var(test) = ~2; xlog("L_INFO", "test is <$var(test)>\n"); and i get to syslog: May 17 18:10:40 sip /usr/sbin/sip-proxy[29280]: INFO: test is <2> have i misunderstood the operator or is there a bug? -- juha

3 5

Re: [SR-Users] IPv6
by Daniel-Constantin Mierla 24 May '11

24 May '11

On 5/20/11 6:33 PM, Jon Farmer wrote: > On 20 May 2011 17:25, Daniel-Constantin Mierla<miconda(a)gmail.com> wrote: > >> the module is good, but not for this case -- the warning seems to be from >> startup, saying that no reverse dns could be done to the ipv6 listen address >> and it is harmless (done for the purpose of building aliases list). >> >> Do you get the warning at runtime? >> >> Normally, the ipv6 should be in between square brackets. >> > Adding the address in question to my hosts file temporarily takes the > warning away :) ok, let me know how is going with the testing of ipv6 -- the support is in core since 2002, but not many entered the challenge so far :-) Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 4

newbie INVITE routing logic help
by Skyler 24 May '11

24 May '11

Hi all, Need some newbie advice on how to write INVITE routing for my scenario. I currently have kamailio+asterisk-realtime integration working in the following way: Outbound calls: USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then Asterisk(1.6.2.18) -> SIP/PSTN This works, though I'd prefer to use Kamailio + LCR and gateway failover as Asterisk doesn't do this well. Here's what I want to do: (for now, small steps) Outbound Calls: USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then Asterisk(1.6.2.18) -> Kamailio(3.1.3)+(lcr) -> SIP/PSTN I know I have to set kamailio as a GW in asterisk instead of the PSTN GW's and add the PSTN GW's to kamailio instead, though I'm at a loss as to how to write the routing logic so that kamailio sends calls from asterisk to PSTN. route[LOCATION] { ...non-INVITE check ends here... if(route(FROMASTERISK)) { # coming from Asterisk --- Does this mean check kamailio local users first before continuing? if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } ## -- If match 11 digits and starts with '1' or if match 10 digits and starts with [2-9] if($rU=~"^(1\d{10})$") || if($rU=~"^([2-9]\d{9})$") { xlog("L_INFO", " FROMASTERISK: This will become route(NORTHAMERICA); \n"); exit; ## to be removed } else { xlog("L_INFO", " FROMASTERISK: This will become route(INTERNATIONAL); \n"); exit; ## to be removed } # new call - send to Asterisk route(TOASTERISK); } Will this logic work? I haven't determined how to write route(NORTHAMERICA) or route(INTERNATIONAL) yet so I'm just logging to debug the logic and syntax. Thanks in advance.

1 0

Kamailio to handle call states
by Yeris Antonio Madrigal Castro 23 May '11

23 May '11

Hi I am planing to change my old T1-E1 pbx, wich is AVAYA, for an asterisk cluster. I am planning this for a call center and I spec 80 currents calls for 30 agents sometimes, so I need to handle big queues and call and agent states, is it possible for kamailio handle the call states and agent states, and leave asterisk for the call center functions? Thanks in advance

1 0

TLS - Kamailio Asterisk - Registration issue
by Lucas Alvarez 23 May '11

23 May '11

Hi, I have confgured Kamailio with TLS in an Kamailio-Asterisk implementation. The phones (Bria - Eyebeam - Aastra 57i) are registering in Kamailio without problems, but the registration is not being forwarded to Asterisk. Kamailio is listening on port tls:5061 only and Asterisk on udp:5080. Should Kamailio has to listen on port 5060 also for UAC module functions? If I enable the port 5060, how do I prevent common SIP - UDP registrations? I mean, I want the registrations to be over TLS only. I would appreciate if someone can put me on the right path. Thanks in advance. Lucas Alvarez

2 1

change ptime problem
by hala alramli 22 May '11

22 May '11

i am trying to route to pstn provider,but i faced a problem . my provider accept ptime 20 while i am send the call with ptime 30 how i can change ptime i have kamailio with freeswitch any help please

4 4

Kamailio + Asterisk 1.6.2 Realtime - REGFWD : 401 not authorized
by Skyler 22 May '11

22 May '11

Hi all, After 3 hours stuck on this I have to ask the group. I am setting up Kam 3.1.3 +Ast 1.6.2.18 + realtime following Daniel's guide on the Asipto site. The problem I see is 401 not authorized when uac tries to register. Below is ngrep . do you see something I am missing or have any pointers? I can't seem to get any devices to register with asterisk. --- TIA interface: any filter: (ip or ip6) and ( port 5060 ) U 2011/05/21 23:40:39.333181 192.168.1.132:5060 -> 192.168.1.104:5060 REGISTER sip:192.168.1.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-cd60fa71. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: testing <sip:4444@192.168.1.104>. Call-ID: 5407498c-e755970e(a)192.168.1.132. CSeq: 55278 REGISTER. Max-Forwards: 70. Authorization: Digest username="4444",realm="192.168.1.104",nonce="Tdiwvk3Yr5JAcX9Iljh/AX0BieRt73Y v",uri="sip:192.168.1.104",algorithm=MD5,response="68bc7ad2de02caa48b8cbb22b 6de7aa6". Contact: testing <sip:4444@192.168.1.132:5060>;expires=30. User-Agent: Linksys/PAP2-3.1.23(LS). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. U 2011/05/21 23:40:39.347248 192.168.1.104:5060 -> 192.168.1.132:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-cd60fa71;rport=5060. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: testing <sip:4444@192.168.1.104>;tag=d0c53e97c845e2d86a9876bb195e3450.e92e. Call-ID: 5407498c-e755970e(a)192.168.1.132. CSeq: 55278 REGISTER. Contact: <sip:4444@192.168.1.132:5060>;expires=60;received="sip:192.168.1.132:5060". Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. U 2011/05/21 23:40:39.349726 192.168.1.104:5060 -> 192.168.1.104:5080 REGISTER sip:192.168.1.104:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK0f62.635b95f3.0. To: sip:4444@192.168.1.104. From: sip:4444@192.168.1.104;tag=a707bfce77c9367d1734afac7b3de6ca-f4de. CSeq: 10 REGISTER. Call-ID: 7f1ddb01-6717(a)192.168.1.104. Content-Length: 0. User-Agent: kamailio (3.1.3 (i386/linux)). Contact: <sip:4444@192.168.1.104:5060>. Expires: 30. U 2011/05/21 23:40:39.351790 192.168.1.104:5080 -> 192.168.1.104:5060 SIP/2.0 401 Unauthorized. v: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK0f62.635b95f3.0;received=192.168.1.104. f: sip:4444@192.168.1.104;tag=a707bfce77c9367d1734afac7b3de6ca-f4de. t: sip:4444@192.168.1.104;tag=as64ca9fe3. i: 7f1ddb01-6717(a)192.168.1.104. CSeq: 10 REGISTER. Server: taridium ipbx. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. k: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.104", nonce="5bad77f5". l: 0. U 2011/05/21 23:40:53.322348 192.168.1.104:5060 -> 192.168.1.132:5060 .... U 2011/05/21 23:40:57.180694 192.168.1.132:5060 -> 192.168.1.104:5060 NOTIFY sip:192.168.1.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-56c97474. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: <sip:192.168.1.104>. Call-ID: dbc8d986-9fbb95e(a)192.168.1.132. CSeq: 47 NOTIFY. Max-Forwards: 70. Event: keep-alive. User-Agent: Linksys/PAP2-3.1.23(LS). Content-Length: 0. U 2011/05/21 23:40:57.187840 192.168.1.104:5060 -> 192.168.1.132:5060 SIP/2.0 200 OK - keepalive. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-56c97474;rport=5060. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: <sip:192.168.1.104>;tag=d0c53e97c845e2d86a9876bb195e3450.e05a. Call-ID: dbc8d986-9fbb95e(a)192.168.1.132. CSeq: 47 NOTIFY. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0.

2 2

how to add RR with advertised_address in front of normal RR?
by Andrew Pogrebennyk 22 May '11

22 May '11

I've been playing with sip load-balancer in EC2 but so far haven't been able to figure out how to insert Record-Route header with advertised_address. AFAIK kamailio rr module adds 2 RR headers (one with the inbound address and one with the outbound address) when your transaction comes with one interface/port and leaves via another. In EC2 the receive socket is the same as send socket and I've had no luck trying to get kamailio insert advertised address as inbound RR and then the outbound RR, by calling record_route_preset() several times in a row too. This is a problem if lb is in front of another proxy. As I see this question has been raised on the mailing lists as far as 6 years ago - I would be surprised if no one has come up with a solution yet. So how to do record-routing in a draft-ietf-sip-record-route-fix AKA rfc5658 style on the machine with one physical interface? -- Sincerely, Andrew Pogrebennyk

3 2

Forwarding client IP to secondary SIP server (asterisk)
by Muhammad Naseer Bhatti 22 May '11

22 May '11

Hello, I am testing Kamailio (OpenSER) as a frontend forwarding calls to asterisk. The setup is working fine and I am able to make call fine. However, Asterisk, naturally, sees calls/connections coming from Kamailio (OpenSER). Is there a way I can forward the client (calle to Kamailio (OpenSER)) IP to asterisk so asterisk can authenticate it? -B

2 1

syntax error?
by Skyler 22 May '11

22 May '11

Hi all, I am attempting to handle NOTIFY within my script but there is a definite error in my syntax. I believe it has to do with $hdr(Event). Anyone know how I should re-write this? if (is_method("NOTIFY") && is_domain_local("$rd") && $hdr(Event) =~ "keep-alive" ) { sl_send_reply("200", "OK - keep-alive"); exit; } TIA

2 2

Re: [SR-Users] syntax error?
by Skyler 22 May '11

22 May '11

Found it.will google more from now on. Sorry. if ( is_method("NOTIFY") && uri==myself && $hdr(Event) =~ "keep-alive" ) { sl_send_reply("200", "OK - keepalive"); xlog("L_INFO", "$hdr(Event) detected - ok sent\n"); exit; } From: Skyler [mailto:skchopperguy@gmail.com] Sent: Saturday, May 21, 2011 7:40 PM To: 'sr-users(a)lists.sip-router.org' Subject: syntax error? Hi all, I am attempting to handle NOTIFY within my script but there is a definite error in my syntax. I believe it has to do with $hdr(Event). Anyone know how I should re-write this? if (is_method("NOTIFY") && is_domain_local("$rd") && $hdr(Event) =~ "keep-alive" ) { sl_send_reply("200", "OK - keep-alive"); exit; } TIA

1 0

Re: [SR-Users] Modules directory missing?
by Skyler 22 May '11

22 May '11

Found it. /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ From: Skyler [mailto:skchopperguy@gmail.com] Sent: Saturday, May 21, 2011 3:29 PM To: 'sr-users(a)lists.sip-router.org' Subject: Modules directory missing? Hello, I am new to Kamailio and have installed via rpm repository on CentOS 5.5 Where is the modules directory located? I cannot load any modules and Kamailio will not start. I've checked these locations but they don't exist. Is there another package I need to install or where did I go wrong? /usr/local/lib/kamailio/modules_k/ /usr/local/lib/kamailio/modules/

1 0

Modules directory missing?
by Skyler 21 May '11

21 May '11

Hello, I am new to Kamailio and have installed via rpm repository on CentOS 5.5 Where is the modules directory located? I cannot load any modules and Kamailio will not start. I've checked these locations but they don't exist. Is there another package I need to install or where did I go wrong? /usr/local/lib/kamailio/modules_k/ /usr/local/lib/kamailio/modules/

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Re: [SR-Users] ACK to 486 is not being proxied
by Carl Wagner 21 May '11

21 May '11

Daniel, Sorry, I had meant to send it to the list. You are correct about missing messages. The ACK is being sent, but it is sent to it's self (loop condition) over the 'lo' device. So basically its a design flaw on my part of the kamailio.cfg file. My current config works for initial messages from the provider and from subscribers. But for initial messages from the Asterisk boxes, I have Asterisk send to the Kamailio box and then have Kamailio look at the message and then route it with a forward(x.x.x.x). While that works for successful calls it is what is breaking the ACK to the 486, and I think it is also breaking Cancel. Do you know of a good example config file where: 1) SIP message from provider hits Kamailio and is routed to one of several Asterisk boxes (load balancing, using ds_select) 2) Messages from a subscriber will be routed to one of the Asterisk boxes (load balancing, using ds_select) 3) Messages from an Asterisk box will be routed to the provider OR a subscribers SIP device based on a fixed string in the R-URI Or you can laugh at what I have: (constructive criticism is welcome from anyone!) (I had to trim this message as it was exceeding the 40K limit) #!KAMAILIO # ####### Defined Values ######### # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 ####### Global Parameters ######### # debug=4 # log_stderror=yes debug=3 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:2.2.2.164:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 dns=no rev_dns=no open_files_limit=4096 disable_core_dump=yes disable_tcp=yes check_via=0 reply_to_via=0 # we may be able to use the above with force_rport ####### Modules Section ######## # set paths to location of modules mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" # loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "dispatcher.so" loadmodule "avpops.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) # the line below puts extra stuf at the end of the log line modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ # ----- usrloc params ----- /* enable DB persistency for location entries */ # appears to use table 'location' and this table has data in it modparam("usrloc", "db_url", "mysql://kamailio:password@my.db.com/kamailio") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # ----- auth_db params ----- # appears to use table 'subscriber' and this table has data in it modparam("auth_db", "db_url", "mysql://kamailio:password@my.db.com/kamailio") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", 1) modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("avpops","db_url","mysql://kamailio:password@my.db.com/AppServer") modparam("avpops","avp_table","User") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # initial checks (check for bad/malformed/unsupported messages) route(REQINIT); # if it is not a OPTIONS message - log it if (!is_method("OPTIONS")) { xlog("L_INFO", "======== processing new message\n"); xlog("L_INFO", "MAIN: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci]) CSeq=[$cs]\n"); } # handle requests within SIP dialogs (active ongoing call - has TO: tag) does not return route(WITHINDLG); ### only initial requests (no To: tag) from here on # CANCEL processing if (is_method("CANCEL")) { # - this needs some work and cancels are not being replied to or forwared # cancle is a hop-by-hop message (send a 200 OK then forward it on) # t_check_trans = if cancel, this is supposed to be true if the Invite transaction exits # see: http://www.kamailio.org/docs/modules/1.1.x/tm.html#AEN460 if (t_check_trans()) { xlog("L_INFO", "MAIN: Recieved Cancel, t_check_trans is true\n"); t_relay(); exit; } xlog("L_INFO", "MAIN: Recieved Cancel, t_check_trans is FALSE\n"); # forward it on (not sure why we are getting here with valid calls that need to be canceled) # this may be bad as somone might be able to "cancel storm" us?? t_relay(); exit; } # this was added to handle OPTIONS messages route(HANDLEOPTIONS); # I 'think' this absorbs retransmissions of Invites t_check_trans(); if (!is_method("REGISTER")) { if ($rU==$null) { # request with no Username in RURI xlog("L_INFO", "Error: main route: request with no Username in RURI\n"); sl_send_reply("484","Address Incomplete"); exit; } } # set up a trigger for failures of INVITEs if (is_method("INVITE")) { xlog("L_INFO", "main_loop: SIP Request: ruri=[$ru] (method [$rm] from [$fu] to [$tu])\n"); t_on_failure("FAIL_ONE"); } # at this point we need to determine how to route the call # if the call is from PROVIDER, it has to go to an AST box (L3 should never go directly to a subscriber) # if the call is from a subscriber, it has to go to an AST box (sub can't dial directly out) # if the call is from an AST box we need to determine if it goes to L3 or a subscriber # if to a subscriber the ToURI should contain "SIP_" # log all invites to syslog # if (is_method=="INVITE")) # { # xlog("L_INFO", "main_loop: SIP Request: ruri=[$ru] (method [$rm] from [$fu] to [$tu])\n"); # }; # handle the easy cases first # from PROVIDER if ($src_ip=="3.3.3.40") { route(FROMPROVIDER); exit; # just to be sure } # if from an AST box (will determine wether to send it to PROVIDER or a subscriber) if ($src_ip=~"2\.2\.2\.1") # regexp match { route(FROMAST); exit; # just to be sure } force_rport(); # attemp to make kamailio use the emhemeral port of the sender #=============== Only subscribers left from here down # handle registrations (from a sip client) route(REGISTER); # authentication for everyone route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } route(FROMSUBSCRIBER); # route(RELAY); xlog("L_INFO", "ERROR - no route chosen \n"); } ################################################################### # Per SIP request initial checks route[REQINIT] { # xlog("L_INFO", " REQINIT: Initial Checks\n"); if (!mf_process_maxfwd_header("10")) { xlog("L_WARN", " REQINIT: Too Many Hops (Request: [$rm])\n"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("L_WARN", " REQINIT: Malformed SIP message from $si:$sp (Request: [$rm]) \n"); exit; } # xlog("L_INFO", " leaving [REQINIT] \n"); # unsupported messages here if (is_method("SUBSCRIBE")) { # probably want to comment the next line out after it is working xlog("L_WARN", "REQINIT: Subscribe not supported: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci]) CSeq=[$cs]\n"); exit; } # if no errors, just fall through and return to caller } ################################################################### # Handle requests within SIP dialogs (request has a TO: Tag) route[WITHINDLG] { # xlog("L_INFO", " WITHINDLG: starting up (WithInDialog)\n"); if (has_totag()) { xlog("L_INFO", " WITHINDLG: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci], CSeq=[$cs])\n"); xlog("L_INFO", " WITHINDLG: has TO: tag\n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is true\n"); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } xlog("L_INFO", " WITHINDLG: sending to relay\n"); route(RELAY); } else # not loose_route { xlog("L_INFO", " WITHINDLG: has TO: tag AND louse_route is NOT true\n"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK\n"); if ( t_check_trans() ) # see if a message is related to a transaction { # Reference: http://www.kamailio.org/docs/modules/stable/modules/tm.html#t_check_trans xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK and t_check_trans=true\n"); # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK and t_check_trans=FALSE\n"); xlog("L_INFO", " WITHINDLG: the ACK to a 486 was not being processed so I am adding t_relay here\n"); # $var(a) = t_relay(); # $var(a) = forward(); t_relay(); xlog ("L_INFO", " WITHINDLG: (ReturnCode = [$var(a)] exiting)\n"); exit; } } # not an ACK - as both branches aove exit xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method not = ACK \n"); sl_send_reply("404","Not here"); } exit; } # xlog("L_INFO", " leaving [WITHINDLG] (WithInDialog)\n"); # if msg has to: tag it should be processed above and then exit, else just return for further processing } ################################################################### # By CEW 2Feb2011 route[REGISTER] { if (is_method("REGISTER")) { # NOTE: need to review: http://www.kamailio.org/docs/scripting.html on how they handle registration # at this point all that should be left are sip clients - they are coming over the open internet so authenticate them! # xlog("L_INFO", " REGISTER: processing register from sip client, fu=$fu \n"); # authenticate the REGISTER requests (uncomment to enable auth) # www_authenticate will send a 401 Unauthorized (and will contain WWW-Authenticate in the reply) # proxy_authenticate will send a 407 Proxy Authentication Required. (and will contain Proxy-Authenticate in the reply) # uses the kamailio->subsriber DB table # if (!proxy_authenticate("2.2.2.164", "subscriber")) if (!www_authenticate("2.2.2.164", "subscriber")) { $var(a) = $rc; # save the return from the www_authenticate # -1 = generic error # -2 = invalid password # -3 = invalid user ( or domain is wrong) if ($var(a) == -2) { xlog ("L_INFO", " REGISTER ERROR: bad password, (ReturnCode = $var(a) exiting)\n"); exit; } if ($var(a) == -3) { xlog ("L_INFO", " REGISTER ERROR: invalid user, (ReturnCode = $var(a) exiting)\n"); exit; } xlog("L_INFO", " REGISTER: issuing a challenge, 401 (and then exit script)\n"); # send a "401 Unauthorized", with realm=2.2.2.164, caller should resend invite with auth info www_challenge("2.2.2.164", "1"); # send a 401 # proxy_challenge("2.2.2.164", "1"); # send a 407 # xlog("L_INFO", " REGISTER: exiting\n"); exit; } xlog("L_INFO", " REGISTER: Match found in database for register\n"); # make sure the Authorization: username matches the To: username # $au = user part of username from Authorization: or Proxy-Authorization header (from the resent Invite, after the challenge) # $tU = reference to username in URI of 'To' header if ($au!=$tU) { xlog("L_INFO", " REGISTER: sending 403 Forbidden auth ID\n"); sl_send_reply("403","Forbidden auth ID"); exit; } # now store it in the DB and then exit if (!save("location")) { xlog("L_INFO", " REGISTER: Error saving location, sending SIP error to other end\n"); sl_reply_error(); } exit; } } ################################################################### # Authentication route route[AUTH] { xlog("L_INFO", " entering route[AUTH] Authentication route\n"); # authenticate if from local subscriber # I think this should be true for LOCAL sip callers (in production there shouldn't be any of these) if (from_uri==myself) { xlog("L_INFO", " AUTH: from_uri = myself ($fu)\n"); if (!proxy_authenticate("$fd", "subscriber")) { $var(a) = $rc; # save the return from the www_authenticate # -1 = generic error - like no valid auth # -2 = invalid password # -3 = invalid user (or domain is wrong) if ($var(a) <= -2) { xlog ("L_INFO", " AUTH: ERROR: Location lookup, ReturnCode = $var(a) exiting\n"); exit; } # xlog("L_INFO", " AUTH: no authenticate credentials supplied, sending challenge (407), from_uri=$fu \n"); xlog("L_INFO", " AUTH: no authenticate credentials supplied, sending challenge (401), from_uri=$fu \n"); # send the proxy challenge and exit. Let the caller send another invite with authentication info. proxy_challenge("$fd", "0"); exit; } xlog("L_INFO", " AUTH: proxy_autorization OK! \n"); # make sure the Authorization: username matches the To: username # $au = user part of username from Authorization: or Proxy-Authorization header (from the resent Invite, after the challenge) # $fU = reference to username in URI of 'From' header if ($au!=$fU) { xlog("L_INFO", " AUTH: Forbidden, sending 403\n"); sl_send_reply("403","Forbidden auth ID"); exit; } xlog("L_INFO", " AUTH: Authorization accepted! Consuming credentials\n"); consume_credentials(); # caller authenticated } else { # !! NOTE: this branch has not been tested as I cant simulate it in my lab!!! # a lot of it was just made up so it does need a lot of testing. # subscriber from the Internet # this is where all our sip clients should hit xlog("L_INFO", " AUTH: EXTERNAL subscriber, need to authenticate from_uri $fu\n"); # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { xlog("L_INFO", " AUTH: NOT RELAYING!, (uri!=myself) sending 403\n"); sl_send_reply("403","Not relaying"); exit; } # not sure if we need to do a proxy_authenticate or www_authenticate here - need to test in production if (!proxy_authenticate("$fd", "subscriber")) { xlog("L_INFO", " AUTH: sending proxy_challenge, from_uri=$fu, exiting \n"); # send the proxy challenge and exit. Let the caller send another invite with authentication info. proxy_challenge("$fd", "0"); xlog("L_INFO", " AUTH: sent proxy_challenge, exiting\n"); exit; } # make sure the Authorization: username matches the To: username # $au = user part of username from Authorization: or Proxy-Authorization header (from the resent Invite, after the challenge) # $fU = reference to username in URI of 'From' header if ($au!=$fU) { xlog("L_INFO", " AUTH: Forbidden auth ID, from_uri=$fu, exiting \n"); sl_send_reply("403","Forbidden auth ID"); exit; } xlog("L_INFO", " AUTH: Consuming credentials\n"); consume_credentials(); } xlog("L_INFO", " leaving route[AUTH] Authentication route\n"); return; } ################################################################### # if the call is from PROVIDER, it has to go to an AST box (L3 should never go directly to a subscriber) # if (uri=="3.3.3.40") route[FROMPROVIDER] { # xlog("L_INFO", " FROMPROVIDER: starting up \n"); if (is_method("REGISTER")) { xlog("L_INFO", " FROMPROVIDER: REGISTER from PROVIDER, sending 200 OK fd=$fd \n"); sl_send_reply("200", "ok"); exit; }; # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); }; # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting }; # if a conference call, send it to ast1 # ds_select_dst(set, alg) the method selects a destination from address set # set - first column from dest file (dispacher.list) # alg 0 = hash over callid if (uri=~"12223334444") { xlog("L_INFO", " FROMPROVIDER: Conference call - sending to ast1 \n"); ds_select_dst("3", "0"); forward(); exit; }; # avp_check(name, op_value) check the value of the avp against an operator and value # op_value = operator '/' value ['/'flags] # $fu = # $ru = SIP Request's URI # if the call came from ast0 (2.2.2.167) via PROVIDER route it to ast1 or we will have a loop condition # $ci = the call ID # AVP = attribute-value-pair if (avp_check("$ci","re/.*(a)2.2.2.167.*/g")) { xlog ("L_INFO", " FROMPROVIDER: Call originating on ast0, sending call to ast1 [$ru]\n"); ds_select_dst("2", "0"); forward(); exit; }; # if the call came from ast1 (2.2.2.168) via PROVIDER route it to ast0 or we will have a loop condition if (avp_check("$ci","re/.*(a)2.2.2.168.*/g")) { xlog ("L_INFO", " FROMPROVIDER: Call originating on ast1, sending call to ast0 [$ru]\n"); ds_select_dst("3", "0"); forward(); exit; }; # Extract the telephone number from the To-URI $avp(i:999)=$tU; # insted of the next line, should we just do a regexp to remove all non-numeric? avp_subst("$avp(i:999)","/\+//g"); # remove the literal string "+" from the variable, if any # xlog("L_INFO", " FROMPROVIDER: about to do \"SELECT AppName FROM Application WHERE TelephoneNumber = $avp(i:999) \" \n"); avp_db_query("SELECT AppName FROM Application WHERE TelephoneNumber = '$avp(i:999)'","$avp(i:888)"); # xlog("L_INFO", " FROMPROVIDER: Query result =[$avp(i:888)] \n"); # see if we got anything back from the DB, if we did it would look like "product" or some future varient if(!is_avp_set("$avp(i:888)")) # if avp 888 is not set, dump the call { xlog ("L_INFO", " FROMPROVIDER: Bad number. Sending 404. method=[$rm] ruri=[$ru] to=[$tu] from=[$fu]\n"); # I don't think the next line is needed as it appears this table is never read (just written to) avp_db_query("INSERT IGNORE INTO InvalidTN VALUES ('$avp(i:999)')"); sl_send_reply("404", "Not Found"); # do we really want to do this? Or shoud we just silently drop it? exit; }; # this is where we do the main load balancing, ast0 or ast1 xlog ("L_INFO", " FROMPROVIDER: Normal Round-Robin call, routing to [$ru], from [$fu]\n"); # ds_select_dst("1", "0"); ds_select_dst("3", "0"); forward(); # Note: forward does return, so we could do post call processing here exit; } ################################################################### # if the call is from an AST box we need to determine if it goes to L3 or a subscriber # if to a subscriber the ToURI should contain "SIP_" # if (uri=~"2\.2\.2\.1") from our ast box route[FROMAST] { xlog("L_INFO", " FROMAST: starting up \n"); if (is_method("REGISTER")) { xlog("L_INFO", " FROMAST: REGISTER from AST, sending 200 OK fd=$fd \n"); sl_send_reply("200", "ok"); exit; }; # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # this block is probalby not needed, except possilby for logging $avp(i:999)=$tU; # Extract the telephone number from the To-URI avp_subst("$avp(i:999)","/\+//g"); # remove the literal string "sip:+" from the variable xlog("L_INFO", " FROMAST: avp(999) = $avp(i:999)\n"); # at this point we need to see if the number contains a "SIP_". If so route it to the subscriber, else route it to L3 xlog("L_INFO", " FROMAST: tU=$tU\n"); if ($tU=~"SIP_") { xlog("L_INFO", " FROMAST: contains \"SIP_\" so routing to subscriber\n"); route(TOSUBSCRIBER); exit; # should be handled in TOSUBSCRIBER, but just to make sure we bail here and don't continue } xlog("L_INFO", " FROMAST: does not contain SIP_ so routing to PROVIDER\n"); # need to rewrite the RURI and To URI $td="3.3.3.40:5060"; $rd="3.3.3.40:5060"; forward("3.3.3.40"); # send it on to PROVIDER } ################################################################### # if the call is from a subscriber, it has to go to an AST box (subscribers can't dial directly out) route[FROMSUBSCRIBER] { xlog("L_INFO", " FROMSUBSCRIBER: starting up \n"); # at this point we should have already been authenticated by route[AUTH] so just route it to an ast box # this is where we do the main load balancing, ast0 or ast1 (round robin) ds_select_dst("3", "0"); forward(); xlog("L_INFO", " leaving route[FROMSUBSCRIBER] \n"); exit; # just in case } ################################################################### # The call needs to be routed to a scubscriber route[TOSUBSCRIBER] { xlog("L_INFO", " TOSUBSCRIBER: starting up \n"); # this is NOT tested yet! # do a DB lookup on kamailio->location to see where to route (should only be for SIP-to-Subscribers) if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); xlog("L_INFO", " TOSUBSCRIBER: Subscriber not Found, sending 404 \n"); t_reply("404", "Not Found"); exit; case -2: xlog("L_INFO", " TOSUBSCRIBER: Method Not allowed, sending 405 \n"); sl_send_reply("405", "Method Not Allowed"); exit; } } xlog("L_INFO", " TOSUBSCRIBER: subscriber is authorized! \n"); # not sure about any of this that follows # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { xlog("L_INFO", " TOSUBSCRIBER: setting flag FLT_ACCMISSED \n"); setflag(FLT_ACCMISSED); } xlog("L_INFO", " TOSUBSCRIBER: attempting to do: t_relay \n"); if (!t_relay()) { xlog("L_INFO", " TOSUBSCRIBER: ERROR in t_relay \n"); sl_reply_error(); }; xlog("L_INFO", " TOSUBSCRIBER: message proxied, now exiting \n"); exit; } ################################################################### # Sample failure route # One source for this block: http://pastebin.com/NMZiBuNm # A source: http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#failure_route # 'failure_route' is executed only by TM module after it was armed via t_on_failure(“failure_route_index”) failure_route[FAIL_ONE] { xlog("L_INFO", " FAIL_ONE: starting up Sample failure route (looks for canceled)\n"); if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { xlog("L_INFO","[1] ($ru) \n"); sl_send_reply("ACK", "Busy"); # send an ACK back to sender t_relay(); # forward it on } # this may do what I want ?? # if(is_method("INVITE")) { xlog("L_INFO", " leaving route[FAIL_ONE] Sample failure route \n"); } ################################################################### route[RELAY] { xlog("L_INFO", " RELAY starting up\n"); /* example how to enable some additional event routes */ if (is_method("INVITE")) { xlog("L_INFO", " RELAY: Error: INVITE sending sl_on_failure\n"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { xlog("L_INFO", " RELAY: Error sending sl_reply_error\n"); sl_reply_error(); } xlog("L_INFO", " leaving [RELAY]\n"); exit; } ################################################################### # Presence server route route[PRESENCE] { xlog("L_INFO", " PRESENCE: starting up \n"); if(!is_method("PUBLISH|SUBSCRIBE")) return; # if presence enabled, this part will not be executed # if (is_method("PUBLISH") || $rU==$null) # { xlog("L_INFO", " PRESENCE: PUBLISH and SUBSCRIBE not supported sending 404\n"); sl_send_reply("404", "Not here"); exit; # } # xlog("L_INFO", "leaving route[PRESENCE] Presence server route\n"); # return; } ################################################################### # Caller NAT detection route route[NAT] { xlog("L_INFO", " NAT: NAT detection route (currently does nothing)\n"); xlog("L_INFO", " leaving route[NAT] NAT detection route \n"); return; } ################################################################### # RTPProxy control route[RTPPROXY] { xlog("L_INFO", " RTPPROXY: RTPProxy control(currently does nothing)\n"); xlog("L_INFO", " leaving route[RTPPROXY] RTPProxy control\n"); return; } ################################################################### # Routing to foreign domains route[SIPOUT] { xlog("L_INFO", " SIPOUT: Routing to foreign domains\n"); if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } xlog("L_INFO", " leaving route[SIPOUT] Routing to foreign domains\n"); } ################################################################### # PSTN GW routing route[PSTN] { xlog("L_INFO", " PSTN: PSTN GW routing (currently does nothing) \n"); xlog("L_INFO", " leaving route[PSTN] PSTN GW routing\n"); return; } ################################################################### # PROVIDER or our AST boxes may send an options message, they are our friends so just 200 OK them # By CEW 2Feb2011 route[HANDLEOPTIONS] { if (is_method("OPTIONS")) { # xlog("L_INFO", " HANDLEOPTIONS: found options message \n"); # if from AST if ($src_ip=~"2\.2\.2\.1") { # xlog("L_INFO", " HANDLEOPTIONS from our AST, sending 200 OK\n"); sl_send_reply("200", "ok"); exit; }; # if from PROVIDER if ($src_ip=~"2\.2\.2\.40") { xlog("L_INFO", " HANDLEOPTIONS from PROVIDER, sending 200 OK\n"); sl_send_reply("200", "ok"); exit; }; # if we made it here, just throw the message away # this assumes that none of our sip clients will send us Options messages! exit; }; }

1 0

Via branch parameter in end2end ACK
by Juha Heinanen 20 May '11

20 May '11

Daniel-Constantin Mierla writes: > if I understood right, it is about 200 ACK which are forwarded > stateless. If yes, then should be only in forward.c, the condition at > line 545, where the assignment can be changed in a memcpy. for example like this? if so, i can go and make the commit. -- juha if (syn_branch ) { memcpy*(*msg->add_to_branch_s, "z9hG4bKcydzigwkX", 16); msg->add_to_branch_len=16; } else {

2 1

IPv6
by Jon Farmer 20 May '11

20 May '11

Hi How do I represent the following if the address was IPv6? if(src_ip ==127.0.0.1) If I specify the IPv6 address with our without [] I get WARNING:core:fix_socket_list: could not rev. resolve <ipv6_addr> Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 2

xHTTP module.
by David J. 20 May '11

20 May '11

I was wondering if I can make an HTTP request to Kamailio; and then have kamailio do a lookup based on passed parameters to connect callers. I am trying to make a click-to-dial type application; I was looking at the HTTP server inside kamailio; It was interesting to me to try to use this as an alternative to php-asterisk; (I know kamailio is not a full blown application server;) If anyone could help me with this I would appreciate it; Thanks in advance. David.

2 4

Re: [SR-Users] ACK to 486 is not being proxied
by Daniel-Constantin Mierla 20 May '11

20 May '11

Please do not write private emails, keep the mailing list cc-ed. Can you get the output of: ngrep -d any -qt -W byline port 5040 on kamailio server for such a call? Include all messages, since you may miss something in your filtering (like you did below), of course you can mask the ip addresses. Also, be sure you do t_relay() to forward the invite from kamailio on. Cheers, Daniel On 5/20/11 3:47 PM, Carl Wagner wrote: > Daniel, > > With a non-2xx reply being hop-by-hop, shouldn't Kamailio receive the > 486, send the ACK to the sender, and then proxy the 486? > > I am not seeing the 486 message in the Kamailio log so I assume that > it automatically handles it without hitting the kamailio.cfg script? > Or do I need to set up a failure_route to catch the 486 and send the > ACK there? > > Please see the log below. > > Thanks, > Carl > > > > # Asterisk to Kamailio > U 2011/05/19 14:13:01.467157 3.4.5.167:5060 -> 3.4.5.164:5060 > INVITE sip:+13031112222@3.4.5.164 SIP/2.0. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@3.4.5.164>. > Contact: <sip:+13032223333@3.4.5.167>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > User-Agent: Asterisk. > Max-Forwards: 70. > Date: Thu, 19 May 2011 20:13:01 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 238. > . > v=0. > o=root 27806 27806 IN IP4 3.4.5.167. > s=session. > c=IN IP4 3.4.5.167. > t=0 0. > m=audio 13424 RTP/AVP 0 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > # Kamailio to provider > U 2011/05/19 14:13:01.468627 3.4.5.164:5060 -> 2.3.4.40:5060 > INVITE sip:+13031112222@2.3.4.40:5060 SIP/2.0. > Record-Route: <sip:3.4.5.164;lr=on>. > Via: SIP/2.0/UDP 3.4.5.164;branch=0. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>. > Contact: <sip:+13032223333@3.4.5.167>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > User-Agent: Asterisk. > Max-Forwards: 69. > Date: Thu, 19 May 2011 20:13:01 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 238. > . > v=0. > o=root 27806 27806 IN IP4 3.4.5.167. > s=session. > c=IN IP4 3.4.5.167. > t=0 0. > m=audio 13424 RTP/AVP 0 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2011/05/19 14:13:01.511775 2.3.4.40:37897 -> 3.4.5.164:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 3.4.5.164;branch=0,SIP/2.0/UDP > 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Content-Length: 0. > . > > > U 2011/05/19 14:13:01.513162 3.4.5.164:5060 -> 3.4.5.167:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Content-Length: 0. > . > > # provider says they are busy > U 2011/05/19 14:13:03.239462 2.3.4.40:37897 -> 3.4.5.164:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 3.4.5.164;branch=0,SIP/2.0/UDP > 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > Record-Route: <sip:3.4.5.164;lr=on>. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>;tag=VPST506071629352. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Contact: <sip:+13031112222@2.3.4.40:5060;transport=udp>. > Content-Length: 0. > . > > # > # should Kamailio send the ACK here? > # > > # Kamailio proxies the 486 to Asterisk > U 2011/05/19 14:13:03.240820 3.4.5.164:5060 -> 3.4.5.167:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > Record-Route: <sip:3.4.5.164;lr=on>. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>;tag=VPST506071629352. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Contact: <sip:+13031112222@2.3.4.40:5060;transport=udp>. > Content-Length: 0. > . > > > # ACK from Asterisk - should be swallowed by Kamailio > U 2011/05/19 14:13:03.243870 3.4.5.167:5060 -> 3.4.5.164:5060 > ACK sip:+13031112222@3.4.5.164 SIP/2.0. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@3.4.5.164>;tag=VPST506071629352. > Contact: <sip:+13032223333@3.4.5.167>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 ACK. > User-Agent: Asterisk. > Max-Forwards: 70. > Content-Length: 0. > . > > > ==== a bunch of retransmitted 486's chopped ============= > > > > > > > > On 05/19/2011 02:30 AM, Daniel-Constantin Mierla wrote: >> Hello, >> >> the ACK for non-2xx replies is hop-by-hop, meaning that kamailio will >> generate one when the reply is received, will send back the reply and >> when the ack comes from upstream, it will absorb it. >> >> In this case it seems that ack is not matching any invite transaction >> and thus is not known where to send it. >> >> Can you send the invite, reply and the ack requests taken with ngrep >> for such case? Maybe is something broken in the content of the message. >> >> Cheers, >> Daniel >> >> On 5/18/11 11:23 PM, Carl Wagner wrote: >>> Hi, >>> >>> I have Kamailio set up to act as a proxy and load balancer. Most >>> things are working correctly but for some reason I can't get >>> Kamailio to proxy an ACK to a 486 Busy. >>> I assume that it is something wrong that I have done in the >>> kamailio.cfg file but I am not seeing it. >>> >>> Does the t_check_trans() return false because the call state is Busy? >>> >>> Please let me know if you need other logs or information. >>> >>> Thanks in advance, >>> Carl >>> >>> >>> >>> ================== Call flow: (from ngrep) ====================== >>> >>> Asterisk Kamailio Provider >>> INVITE---------> >>> INVITE---------> >>> <----------100 Trying >>> <---------100 Trying >>> <----------486 Busy >>> <---------486 Busy >>> ACK-------------> >>> ====== Kamailio does not proxy the ACK here ======= >>> <----------486 Busy >>> <---------486 Busy >>> ACK-------------> >>> <----------486 Busy >>> <---------486 Busy >>> >>> >>> === kamailio.cfg snippet >>> ... >>> route >>> { >>> route(REQINIT); # remove malformed messages >>> >>> # handle requests within SIP dialogs >>> route(WITHINDLG); >>> ... >>> } >>> >>> ################################################################### >>> # Handle requests within SIP dialogs (request has a TO: Tag) >>> route[WITHINDLG] >>> { >>> if (has_totag()) >>> { >>> xlog("L_INFO", " WITHINDLG: SIP Request: [$rm] ruri=[$ru] (from >>> [$fu] to [$tu], Call-ID=[$ci], CSeq=[$cs])\n"); >>> if (loose_route()) >>> { >>> ... >>> } >>> else # not loose_route >>> { >>> if ( is_method("ACK") ) >>> { >>> if ( t_check_trans() ) # see if a message is related to >>> a transaction >>> { >>> ... >>> } >>> else >>> { >>> # ACK without matching transaction ... ignore and >>> discard >>> xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true >>> and is_method = ACK and t_check_trans=FALSE\n"); >>> >>> # not forwarded here!!! Tried both t_relay and forward. >>> # $var(a) = t_relay(); >>> $var(a) = forward(); >>> xlog ("L_INFO", " WITHINDLG: (ReturnCode = [$var(a)] >>> exiting)\n"); >>> >>> exit; >>> } >>> } >>> sl_send_reply("404","Not here"); >>> } >>> exit; >>> } >>> } >>> >>> ============= End of kamailio.cfg ================ >>> >>> >>> ============= /var/log/messages - snippit of the message ========== >>> >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> ======== processing new message >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> MAIN: SIP Request: [ACK] ruri=[sip:+13031112222@2.3.4.5] (from >>> [sip:+13031112222@3.4.5.6] to [sip:+13031112222@2.3.4.5], >>> Call-ID=[21a30b9168e0a8c963d47ca1361cef77(a)3.4.5.6]) CSeq=[102] >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: SIP Request: [ACK] ruri=[sip:+13031112222@2.3.4.5] (from >>> [sip:+13031112222@3.4.5.6] to [sip:+13031112222@2.3.4.5], >>> Call-ID=[21a30b9168e0a8c963d47ca1361cef77(a)3.4.5.6], CSeq=[102]) >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: has TO: tag >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = >>> ACK and t_check_trans=FALSE >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: the ACK to a 486 was not being processed so I am >>> adding t_relay here >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: (ReturnCode = [1] exiting) >>> >>> >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

nonce reuse parameter
by Bruno Bresciani 20 May '11

20 May '11

Hi, on kamailio 3.1.2 the one_time_nonce parameter of the auth module is similar to the nonce_reuse parameter present on kamailio 1.5.0? I want to disable the check of the nonce reuse in kamailio 3.1.2... regards

2 2

Reactivating a failed dispatcher destination
by Lee Archer 20 May '11

20 May '11

Hi all I wonder if someone can help by pointing me in the right direction. I have dispatcher set to ping and set any failed gateways to probing but how do you get them out of this state when they start working again? I have run some tests and I must be missing something as active servers seem to stay in probing mode. I am using Kamailio 3.0.4. Thanks Lee

3 10

packaging 3.1.4
by Daniel-Constantin Mierla 20 May '11

20 May '11

Hello, I am thinking of packaging v3.1.4 next week on Thursday -- last minor release was more than 1.5month ago. Any comments against it? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Forward on unanswered call
by Andrew O. Zhukov 20 May '11

20 May '11

Right now I'm try to add the subj feature. Linksys/SPA2102-3.3.6 (just from the box) respond SIP/2.0 480 Temporarily not available Landline over Asterisk with E1: SIP/2.0 603 Limit exceeded Ether I so unlucky, or just forget it and do not provide it to the customers? If so, notes to developers let's take off 408 code processing from sample config. Thanks in advance for your opinion. Andrew.

2 1

Via branch parameter in end2end ACK
by Francesco Castellano 19 May '11

19 May '11

Dear sirs, using version 3.1.3 (kamailio flavour), I noticed that the Via inserted by the proxy, relaying an end2end ACK, contains a branch parameter equal to 0 (with no magic cookie prefixed). No misbehaviour followed, but I was wonder about the reason for this behaviour and whether it can be a fault of my config: route { [some checks] if (!is_method("REGISTER|MESSAGE|PUBLISH")) { record_route(); } else if (is_method("REGISTER")) { route(HANDLE_REGISTER); exit; } if (loose_route()){ if (!has_totag()){ if (!is_method("ACK")){ send_reply("403", "Preset Route Set Not Allowed"); } exit; } t_check_trans(); if (is_method("ACK")){ if ($du == $null){ handle_ruri_alias(); } if (isflagset(T_UAC_NATED) && isflagset(T_UAC_USER)){ add_contact_alias(); } remove_hf("Proxy-Authorization"); t_relay(); exit; } } [other processing] } Here's an example for the forwarded ACK: U 2011/05/17 15:05:49.392201 proxy_1_ip:5060 -> location_callee_ip:5060 ACK sip:5303838@192.168.130.169;user=phone SIP/2.0. Record-Route: <sip:proxy_1_ip;lr=on;ftag=a2z0myc9r6>. Via: SIP/2.0/UDP proxy_1_ip;branch=0. Via: SIP/2.0/UDP proxy_0_ip;rport=5060;branch=0. Via: SIP/2.0/UDP 192.168.130.171:2048;received=location_caller_ip;branch=z9hG4bK-pud4cfiz8vsh;rport=4791. From: "cento tondo" <sip:100@proxy_0_hostname>;tag=a2z0myc9r6. To: <sip:5375485@proxy_1_hostname;user=phone>;tag=00221b79b1b28969. Call-ID: 3c352c0b8419-9tbq023tzfkp. CSeq: 2 ACK. Max-Forwards: 68. Contact: <sip:100@192.168.130.171:2048;alias=location_caller_ip~4791~1;line=gwwqwi25>;reg-id=1. Content-Length: 0. . where proxy_1 does force_rport and record_route() on ACK too, and proxy_0 does not. Both proxies run the same version, and a similar ACK script processing. Best regards, Francesco Castellano

3 5

ACK to 486 is not being proxied
by Carl Wagner 19 May '11

19 May '11

Hi, I have Kamailio set up to act as a proxy and load balancer. Most things are working correctly but for some reason I can't get Kamailio to proxy an ACK to a 486 Busy. I assume that it is something wrong that I have done in the kamailio.cfg file but I am not seeing it. Does the t_check_trans() return false because the call state is Busy? Please let me know if you need other logs or information. Thanks in advance, Carl ================== Call flow: (from ngrep) ====================== Asterisk Kamailio Provider INVITE---------> INVITE---------> <----------100 Trying <---------100 Trying <----------486 Busy <---------486 Busy ACK-------------> ====== Kamailio does not proxy the ACK here ======= <----------486 Busy <---------486 Busy ACK-------------> <----------486 Busy <---------486 Busy === kamailio.cfg snippet ... route { route(REQINIT); # remove malformed messages # handle requests within SIP dialogs route(WITHINDLG); ... } ################################################################### # Handle requests within SIP dialogs (request has a TO: Tag) route[WITHINDLG] { if (has_totag()) { xlog("L_INFO", " WITHINDLG: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci], CSeq=[$cs])\n"); if (loose_route()) { ... } else # not loose_route { if ( is_method("ACK") ) { if ( t_check_trans() ) # see if a message is related to a transaction { ... } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK and t_check_trans=FALSE\n"); # not forwarded here!!! Tried both t_relay and forward. # $var(a) = t_relay(); $var(a) = forward(); xlog ("L_INFO", " WITHINDLG: (ReturnCode = [$var(a)] exiting)\n"); exit; } } sl_send_reply("404","Not here"); } exit; } } ============= End of kamailio.cfg ================ ============= /var/log/messages - snippit of the message ========== May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: ======== processing new message May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: MAIN: SIP Request: [ACK] ruri=[sip:+13031112222@2.3.4.5] (from [sip:+13031112222@3.4.5.6] to [sip:+13031112222@2.3.4.5], Call-ID=[21a30b9168e0a8c963d47ca1361cef77(a)3.4.5.6]) CSeq=[102] May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: WITHINDLG: SIP Request: [ACK] ruri=[sip:+13031112222@2.3.4.5] (from [sip:+13031112222@3.4.5.6] to [sip:+13031112222@2.3.4.5], Call-ID=[21a30b9168e0a8c963d47ca1361cef77(a)3.4.5.6], CSeq=[102]) May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: WITHINDLG: has TO: tag May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK and t_check_trans=FALSE May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: WITHINDLG: the ACK to a 486 was not being processed so I am adding t_relay here May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: WITHINDLG: (ReturnCode = [1] exiting)

2 1

redirect implementation with dispatcher
by alex pappas 19 May '11

19 May '11

Dear all, I'm trying to create a service which when a call reach Kamailio, this call get's a number of gateways in round robin mode and from them I create uri and append_branches. The all scenario works fine except that I don't get from Dispatcher the round robin mode. Is is the way I ask Dispatcher and i don't get the results I need. In other scenarios Dispatcher works fine. The other thing is that I tried the random alg of Dispatcher and that works. I'm trying to understand what I'm doing wrong and why random alg works but round robin not. I'm attaching my code if anyone has an idea of what I'm doing wrong. Thank you Alex *Scenario description:* * Call enter --> Dialplan (Here if has a match get the attributes which is a list of dispatcher gw group in the form of 100.101.#) --> Dispatcher (For every dispatcher gw group appnd the branch) --> Redirect the call* route[TEST] { if(dp_translate("$avp(s:lcr_id)", "$rU/$rU")) { $var(i) = 0; while($(var(dpattrs){s.select,$var(i),.})!="#") *#attr example is gw group of disptcher in order 100.101.# or in my scenario is 200.# which in Dispatcher is 2 gw* *10.0.0.2 and 10.0.0.3* { $avp(s:disp_dstgrp) = $(var(dpattrs){s.select,$var(i),.}{ s.int}); if(ds_select_domain("$avp(s:disp_dstgrp)", "4")) { append_branch(); revert_uri(); $avp(s:disp_dstgrp) = "null"; } $var(i) = $var(i) + 1; # for the while loop } ####################### REDIRECTION ##################### sl_send_reply("300", "Multiple Choice"); ######################################################### exit; } else { xlog(" ----- No routing found for this call -----"); } }

2 3

Kamailio 1.5 acc on glare BYE
by Tobias Lindgren 18 May '11

18 May '11

Hi all! I have a short question with regards to how the acc module handles (or rather logs) BYEs. As I understand, a BYE followed by an 200 OK will be logged by Kamailio's acc module. This works perfectly fine in my set up. However, one of the carriers that I'm using sometimes respond with a "487 Request Terminated" to BYE messages, stating that "Glare Bye condition encountered". I'm not really sure why this occurs at my carriers', but whenever this happens, acc does not log the BYE message. To my question; what happens in the acc module if sip_code > 200 is received? And, should these BYE's be logged, even though the code is 487? I'm on Kamailio 1.5 with acc logging to MySQL. Br, /Tobias

2 2

403 Proxy forbidden replay for invite that send sip gateway
by hala alramli 18 May '11

18 May '11

Hello every body,could you help me ? i bought Geographical DID numbers from gateway provider , to receive calls from pstn my problem that when gateway send invite my proxy send 403 forbidden ,i dont know why what is wrong ,any help please. this is a track for th gateway invite. my proxy ip:79.12.254.153gateway IP :77.239.128.10my geographical number :0698962030 INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44 From: <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 To: <sip:0698962030@10.39.6.207:5060;user=phone> Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 CSeq: 1 INVITE User-Agent: CS2000_NGSS/8.0 Remote-Party-ID: <sip:0289089500@10.39.1.16;user=phone>; party=calling; privacy=off; screen=yes Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK P-Asserted-Identity: <sip:0289089500@10.39.1.16;user=phone> Max-Forwards: 69 Contact: <sip:77.239.128.10:5060;transport=UDP> Supported: 100rel Content-Type: application/sdp Content-Length: 236 v=0 o=PVG 1305215180830 1305215180830 IN IP4 77.239.128.10 s=- p=+1 6135555555 c=IN IP4 77.239.128.10 t=0 0 m=audio 49264 RTP/AVP 18 8 0 13 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no SIP/2.0 403 Proxy forbidden Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 From: <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 To: <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 CSeq: 1 INVITE Server: Kamailio (1.4.4-notls (x86_64/linux)) Content-Length: 0 ACK sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 From: <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 To: <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 CSeq: 1 ACK Max-Forwards: 69 Content-Length: 0 INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2 From: <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad To: <sip:0698962030@10.39.6.207:5060;user=phone> Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 CSeq: 1 INVITE User-Agent: CS2000_NGSS/8.0 Remote-Party-ID: <sip:026683011@10.39.1.16;user=phone>; party=calling; privacy=off; screen=yes Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK P-Asserted-Identity: <sip:026683011@10.39.1.16;user=phone> Max-Forwards: 69 Contact: <sip:77.239.128.10:5060;transport=UDP> Supported: 100rel Content-Type: application/sdp Content-Length: 236 v=0 o=PVG 1305215458700 1305215458700 IN IP4 77.239.128.10 s=- p=+1 6135555555 c=IN IP4 77.239.128.10 t=0 0 m=audio 41892 RTP/AVP 18 8 0 13 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no SIP/2.0 403 Proxy forbidden Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 From: <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad To: <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 CSeq: 1 INVITE Server: Kamailio (1.4.4-notls (x86_64/linux)) Content-Length: 0 sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 From: <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad To: <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 CSeq: 1 ACK Max-Forwards: 69 Content-Length: 0

2 2

setbflag() documentation missing?
by Juha Heinanen 17 May '11

17 May '11

setbflag() is not documented in core cookbook. is it missing or somewhere else? -- juha

2 2

report from linuxtag, presentations
by Henning Westerholt 17 May '11

17 May '11

Hello, just to give you a quick feedback about the this year linuxtag - it was overall a really nice event in my opinion. A lot of people passed by at our booth to discuss about Kamailio and VoIP in general. Carsten Bock presented about our new IMS extensions, Andreas Granig showed a demo of their new kamailio based carrier product. I hope that we can provide some photos of the event soon. We've did two presentations, one about "Linux at 1&1" including a Kamailio part on wednesday, and Carsten did two "FastForward" talks on wednesday and thursday. I've uploaded the slides to our webserver: http://www.kamailio.org/events/2011-linuxtag/linux-at-1and1.pdf http://www.kamailio.org/events/2011-linuxtag/kamailio-quick.pdf Also interesting, the German Federal Office for Information Security (BSI) presented the upcoming "Gemeinschaft" PBX project version 4 that will use Kamailio (and Freeswitch). They sponsor the project, it aims to provide a secure, scalable and easy to use open source PBX plattform. Best regards, Henning

1 0

Kamailio 3.1.3 uncontrolled core dump
by Dominguez Jover, Ricardo 17 May '11

17 May '11

Hi everybody. As posted in http://lists.sip-router.org/pipermail/sr-users/2011-May/068514.html we are having an uncontrolled core dump. It seems that the crash occurs when some users registers to Kamailio. After upgrading to Kamilio release 3.1.3 we are able to generate the following GDB: GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-32.el5_6.2) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "i386-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/local/kamailio-3.1.3/sbin/kamailio...done. Reading symbols from /lib/libdl.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libdl.so.2 Reading symbols from /lib/libresolv.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libresolv.so.2 Reading symbols from /lib/libc.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libc.so.6 Reading symbols from /lib/ld-linux.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/db_mysql.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/db_mysql.so Reading symbols from /usr/lib/mysql/libmysqlclient.so.15...(no debugging symbols found)...done. Loaded symbols for /usr/lib/mysql/libmysqlclient.so.15 Reading symbols from /usr/lib/libz.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libz.so.1 Reading symbols from /lib/libcrypt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libcrypt.so.1 Reading symbols from /lib/libnsl.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libnsl.so.1 Reading symbols from /lib/libm.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libm.so.6 Reading symbols from /lib/libssl.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libssl.so.6 Reading symbols from /lib/libcrypto.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libcrypto.so.6 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb2.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb2.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb1.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb1.so.1 Reading symbols from /usr/lib/libgssapi_krb5.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libgssapi_krb5.so.2 Reading symbols from /usr/lib/libkrb5.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libkrb5.so.3 Reading symbols from /lib/libcom_err.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libcom_err.so.2 Reading symbols from /usr/lib/libk5crypto.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libk5crypto.so.3 Reading symbols from /usr/lib/libkrb5support.so.0...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libkrb5support.so.0 Reading symbols from /lib/libkeyutils.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libkeyutils.so.1 Reading symbols from /lib/libselinux.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libselinux.so.1 Reading symbols from /lib/libsepol.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libsepol.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/mi_fifo.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/mi_fifo.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libkmi.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libkmi.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/kex.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/kex.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libkcore.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libkcore.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/tm.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/tm.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/tmx.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/tmx.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/sl.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/sl.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rr.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rr.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/pv.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/pv.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/maxfwd.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/maxfwd.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/usrloc.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/usrloc.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/registrar.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/registrar.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/textops.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/textops.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/siputils.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/siputils.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/xlog.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/xlog.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/sanity.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/sanity.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/ctl.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/ctl.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/mi_rpc.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/mi_rpc.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/acc.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/acc.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/auth.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/auth.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/auth_db.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/auth_db.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/permissions.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/permissions.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/alias_db.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/alias_db.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/domain.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/domain.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/nathelper.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/nathelper.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rtpproxy.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rtpproxy.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/tls.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/tls.so Reading symbols from /lib/libnss_files.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libnss_files.so.2 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/sqlops.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/sqlops.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/group.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/group.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/dialplan.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/dialplan.so Reading symbols from /lib/libpcre.so.0...(no debugging symbols found)...done. Loaded symbols for /lib/libpcre.so.0 Reading symbols from /lib/libnss_dns.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libnss_dns.so.2 Core was generated by `/usr/local/kamailio-3.1.3/sbin/kamailio -P /var/run/kamailio.pid -m 64 -u kamai'. Program terminated with signal 11, Segmentation fault. #0 pv_set_ruri_user (msg=0x83622c8, param=0x8320078, op=254, val=0xbfbaa83c) at pv_core.c:1756 1756 val->rs.s[val->rs.len] = '\0'; (gdb) (gdb) (gdb) exit (gdb) quit Last debug messages before the crash are: May 13 09:32:06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "@192.168.64.36" May 13 09:32:06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: ERROR: tm [ut.h:318]: failed to resolve "@192.168.64.36" May 13 09:32:06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches May 13 09:32:06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "@192.168.64.36" May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]: ERROR: tm [ut.h:318]: failed to resolve "@192.168.64.36" May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]: ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) May 13 09:32:16 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 21 May 13 09:32:16 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: ERROR: <core> [tcp_main.c:4315]: connect 192.168.67.21:49320 failed (timeout) May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 21 May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 21 in the hash(-1, 0, 0xb40a02e4) flags 0) May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 23 in the hash(-1, 0, 0x83657ec) flags 0) May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 36 on 23: Broken pipe (32) May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: ERROR: <core> [tcp_main.c:3917]: ERROR: send2child: send_fd failed for 0xb421c7e4 (flags 0x4018), fd 36 May 13 09:32:16 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3027]: ALERT: <core> [main.c:741]: child process 3043 exited by a signal 11 May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3027]: ALERT: <core> [main.c:744]: core was generated May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3027]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3046]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3045]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3039]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3035]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3034]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3042]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3041]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3040]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3038]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3037]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:24 tip1 hb_standby[3440]: Going standby [foreign]. May 13 09:32:25 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3027]: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/ser_ctl: Operation not permitted (1) May 13 09:32:25 tip1 heartbeat: [2048]: info: tip1.umh.es wants to go standby [foreign] May 13 09:32:36 tip1 heartbeat: [2048]: WARN: No reply to standby request. Standby request cancelled. Any help would be appreciated. Cheers, Ricardo Dominguez

2 3

Does t_newtran() behave different than in Kamailio 1.5?
by Iñaki Baz Castillo 17 May '11

17 May '11

Hi, as Kamailio 1.5 TM documentation states, changes to transaction (adding headers, filling AVP's and so) after calling t_newtran() are not saved into the transaction: http://kamailio.org/docs/modules/1.5.x/tm.html#id2510920 NOTE that the changes on the request that are made after this function call will not be saved into transaction!!! However in 3.X TM documentation such note is not present: http://kamailio.org/docs/modules/devel/modules/tm.html#t_newtran So, does 3.X behave different and changes after t_newtran are added to the transaction or not? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 8

core in dialog module
by Anton Roman 17 May '11

17 May '11

Hi all, we got a core in dialog module. We are using kamailio 3.1.2. Below you can find a full backtrace from the dump and the Kamailio compilation options. Please, if you need further information don't hesitate to ask me for it. I can't precise the situation when it is generated because we have a quite high load in this server. Thanks in advance. Antón (gdb) bt full #0 unref_dlg (dlg=0x7f08a9f67da8, cnt=1) at dlg_hash.c:598 d_entry = (struct dlg_entry *) 0x7f10304b8b68 #1 0x00007f08ce92fa02 in run_trans_callbacks_internal (cb_lst=0x7f08aa203e98, type=32768, trans=0x7f08aa203e28, params=0x7fff49059a10) at t_hooks.c:290 cbp = (struct tm_callback *) 0x7f08a9f6e7e0 backup_from = (avp_list_t *) 0x8b3330 backup_to = (avp_list_t *) 0x8b3338 backup_dom_from = (avp_list_t *) 0x8b3340 backup_dom_to = (avp_list_t *) 0x8b3348 backup_uri_from = (avp_list_t *) 0x8b3320 backup_uri_to = (avp_list_t *) 0x8b3328 #2 0x00007f08ce92fc56 in run_trans_callbacks (type=32768, trans=<value optimized out>, req=0x1, rpl=0x7f10304b8b68, code=-868566200) at t_hooks.c:317 params = {req = 0x0, rpl = 0x0, param = 0x7f08a9f6e7f0, code = 0, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = { s = 0x0, len = 0}} #3 0x00007f08ce915b36 in free_cell (dead_cell=0x7f08aa203e28) at h_table.c:136 b = <value optimized out> i = <value optimized out> rpl = <value optimized out> tt = <value optimized out> foo = <value optimized out> cbs = <value optimized out> ---Type <return> to continue, or q <return> to quit--- __FUNCTION__ = "free_cell" #4 0x00007f08ce9319f1 in wait_handler (ti=<value optimized out>, wait_tl=<value optimized out>, data=<value optimized out>) at timer.c:645 p_cell = (struct cell *) 0x7f08aa203e28 #5 0x0000000000513d8f in timer_main () at timer.c:894 No locals. #6 0x000000000046501b in main_loop () at main.c:1618 i = 4 pid = <value optimized out> si = (struct socket_info *) 0x0 si_desc = "udp receiver child=3 sock=XXX.XXX.XXX.XX:XXXX\000\000\000\210�\231\000\000\000\000\000\031", '\0' <repeats 15 times>, "\001\000\000\000\000\000\000\000�\215\213", '\0' <repeats 13 times>, "\004", '\0' <repeats 15 times>, "\b\236\005I�\177\000\000\227%J\000\000\000\000" #7 0x0000000000467873 in main (argc=<value optimized out>, argv=0x7fff49059e08) at main.c:2398 cfg_stream = (FILE *) 0x12e1010 c = <value optimized out> r = <value optimized out> tmp = 0x7fff4905ae90 "" tmp_len = 32520 port = <value optimized out> proto = <value optimized out> ret = <value optimized out> seed = 1235801225 ---Type <return> to continue, or q <return> to quit--- rfd = 4 debug_save = <value optimized out> debug_flag = 0 dont_fork_cnt = 0 n_lst = <value optimized out> p = <value optimized out> (gdb) (gdb) quit kamailio2:/var/kamailio# kamailio -V version: kamailio 3.1.2 (x86_64/linux) eb24c1-dirty flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: eb24c1 -dirty compiled on 09:35:52 Apr 28 2011 with gcc 4.3.2

2 5

geoip problem, finally solved
by Francesco Castellano 16 May '11

16 May '11

Dear Sirs, I found some problems using the geoip module. My system is a debian squeeze x86_64; and I installed the following packages: Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Inst/Conf-files/Unpacked/halF-conf/Half-inst/trig-aWait/Trig-pend |/ Err?=(none)/Reinst-required (Status,Err: uppercase=bad) ||/ Nome Versione Descrizione +++-=============================================-=============================================-========================================================================================================== ii geoip-bin 1.4.7~beta6+dfsg-1 IP lookup command line tools that use the GeoIP library ii geoip-database 1.4.7~beta6+dfsg-1 IP lookup command line tools that use the GeoIP library (country database) ii libgeoip-dev 1.4.7~beta6+dfsg-1 Development files for the GeoIP library ii libgeoip1 1.4.7~beta6+dfsg-1 A non-DNS IP-to-country resolver library Using /usr/share/GeoIP/GeoIP.dat as database, the geoip_match("$si", "src") function failed with retcode -2 (NULL from GeoIP_record_by_name()). And finally I found the problem with the GeoIP file DB itself. In particular, compiling a mini program for debug, it said: Invalid database type GeoIP Country Edition, expected GeoIP City Edition, Rev 1 After manually downloading the GeoLiteCity.dat from MaxMind.com everything went fine. I wrote for letting others know my solution, and for asking you if there is a cleaner way to solve it. Best regards, Francesco Castellano

2 3

nathelpler module path support
by Asgaroth 16 May '11

16 May '11

Hi All, I have a scenario where our registrar's are sitting behind our sip proxy. The sip proxy is the only system that has access to the outside world. Our registrar's are all on internal private addresses. I am trying to configure our registrar's to perform nat pinging. However, because the registrar's are on an internal address there is no way for the nat ping to get out. I enabled the path option so that the registrar stores the path that the registration comes in on. However it doesnt seem like the nathelper module uses the path parameter to send the outgoing nat ping. I tried the nat ping using sip method INFO, but the same applies, it does not appear to be able to send out via the outbound proxy. I tried to enable the natping option on the sip proxy, but it appears that only registrations with a socket address of the local host will get pinged, or else it is ignored. Is there an option that I am missing for the nat helper module that will enable me to set an outbound proxy for the nat ping messages? Any insight would be appreciated. Thanks

3 6

LCR strip and tags (prefix) - Kamailio 3.1
by Alexandre Abreu 14 May '11

14 May '11

Hi, With this LCR release from Kamailio 3.1 the table 'lcr_gw' makes four fields unique: lcr_id, ip_addr, port and hostname. This table is also responsible for setting 'strip' and 'tag' parameters values for the gateways. What I am trying to accomplish is to use different strip and tag values for the SAME gateway based only on what was dialed by the user. In 1.5 LCR for example, if user dialed 444 or 555 and both prefix belongs to the same gateway, I could use gw-sip for 444 (with values strip 1 and prefix 22) and also gw-sip for 555 (with values strip 2 and prefix 33). While this duplicate data, it was possible to play with routes without touching the config file. I am looking for ideas on how to implement that on K3.1 without using ifs, strip and prefix tied (statically) in the script file. As far as I know, for every new lcr_id inserted, I will need a corresponding load_gws(new_lcr_id) in the script. Regards, Alexandre.

2 1

DBTEXT - authentication against text files
by Pavel Segeč 13 May '11

13 May '11

Hi, does anybody have some knowledge or experience with DBTEXT module? I'm searching for working DBTEXT files examples and config files examples... thanks palo

1 0

[OT] SIP presence and BLF testing tools
by Andrew Pogrebennyk 12 May '11

12 May '11

Is there any framework available to do functional and regression testing of SIP presence and BLF? Ideally I'm looking for a tool for asserting the signaling flow and contents of the XML body (state, version numbers etc), also I'd like to be able to extract the tags and call-id into variables so as to create INVITE with Replaces. Working with message body variables in sipp is a pain. I'm also familiar with SIPr, Net::SIP and a few others, but that doesn't look suitable for someone without much development experience. Spirent is a perhaps a little closer than others in implementing pickup scenario but awkward when you go beyond the default test scenarios. So I'm interested in any information, tips, suggestions, commercial tools etc. I'm asking here 'cause judging from the quality of their BLF&presence OpenSIPS and Kamailio got the testing process just right. -- Sincerely, Andrew Pogrebennyk

2 1

core dump
by Dominguez Jover, Ricardo 12 May '11

12 May '11

Hi everybody, We are having an uncontrolled crash in Kamailio 3.1: May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23366]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 39 May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT: <core> [main.c:741]: child process 23364 exited by a signal 11 May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT: <core> [main.c:744]: core was not generated I've read in several posts I must generate the "Core Dump" to know what is happening. I'm trying to generate it in this way. "/etc/init.d/Kamailio" file: if test "$DUMP_CORE" = "yes" ; then 1. set proper ulimit ulimit -c unlimited directory for the core dump files COREDIR=/dumps/ [ -d $COREDIR ] || mkdir $COREDIR chmod 777 $COREDIR echo "$COREDIR/core.%e.sig%s.%p" > /proc/sys/kernel/core_pattern fi "/etc/default/Kamailio" file: DUMP_CORE=yes But the core is not yet generated. Then I've added -w option: OPTIONS="-P $PID_FILE -m $MEMORY -u $USER -g $GROUP -w /dumps/" But core still not generated In "kamailio.cfg" I´ve also added: disable_core_dump=no User running Kamailio is "kamailio" who has 777 permissions in /dumps/ directory. No way. Anyhelp would be appreciated. Cheers, Ricardo Domínguez

3 5

Monitor SIP Trunk Usage with MRTG
by JR Richardson 11 May '11

11 May '11

Hi All, So I'm still banging away at monitoring various SIP trunk capacity, the dialog module is working well for this purpose. I am running local server cron job scripts to pull data and put into local flat files for reference. My next step is to pull this data from a remote server and graph with MRTG (this is what I use for most other network monitoring). It looks like I can use the mi_xmlrpc module and remotely query the SIP Router with kamctl commands and parse the results into MRTG. I'm interested in hearing from others that may have already done this and would not mind sharing some scripts or from anyone that has an alternative method. Any guidance or affirmation that this is a good method that works would be helpful. On the MRTG server, I'm using perl scripts for a lot of my data pulling. I am using Kamailio 3.0. Thanks. JR -- JR Richardson Engineering for the Masses

2 3

Kamailio + RTP Proxy modification -- programmer needed
by Salauddin Rubel 11 May '11

11 May '11

Hi, We need a programmer who can modify SIP and RTP packets passing through Kamailio and RTP proxy. Interested programmer can contact with me. Thanks

3 3

Dialog Module Question inbound/outbound count
by JR Richardson 10 May '11

10 May '11

Hi All, Been playing around with the dialog module to help me keep track of SIP trunk utilization, trying to identify how many calls are active on each SIP trunk. My dialog count for inbound calls is 10 (correct) but my outbound calls are always double that and I'm not sure why. Here is a snip of my config: loadmodule "dialog.so" if($rU=~"^111") { dlg_manage(); dlg_setflag("4"); set_dlg_profile("inbound","trunk111"); route(PDT); exit; } PDT lookup for prefix then call is forwarded to dispatcher to route out to carrier through 'route[relay]' route[RELAY] { dlg_setflag("4"); set_dlg_profile("outbound","$rd"); if (!t_relay()) { sl_reply_error(); exit; } exit; } I setup 10 active calls and here are my results: test-sip-router2:~# kamctl fifo profile_get_size inbound profile:: name=inbound value= count=10 test-sip-router2:~# kamctl fifo profile_get_size outbound profile:: name=outbound value= count=20 Also 'kamctl fifo dlg_list' will list only 10 active dialogs. So my question is, why am I getting a 20 count for outbound and only 10 count for inbound? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

t_reply MI Datagram Function
by Torge Szczepanek 10 May '11

10 May '11

Hi! In the Documentation of the TMX Module for the t_reply function there is given a list of Parameters as can be seen here: http://www.kamailio.org/docs/modules/3.1.x/modules_k/tmx.html#id2822345 The third parameter is a trans_id. I cannot find any references for this. Is it possible to send a query to the MI Datagram socket, to do a lookup of the trans_id for a given callid or is there any other solution to get this information through the MI datagram socket, to be able to execute t_reply? It seems that t_reply_callid is doing what I want to do, but currently not through the MI Datagram socket and not including a body or new headers. Greetings Torge

2 2

INSTALL file
by Claudio Furrer 10 May '11

10 May '11

Hello, Is this still applicable? See INSTALL file (line 669) 3.1) create MySQL tables for SER flavour - if you have a previously installed SER on your system, use /usr/sbin/ser_mysql.sh reinstall to convert your SER database into new structures - otherwise, if this is your very first installation, use /usr/sbin/ser_mysql.sh create to create SER database structures (you will be prompted for password of MySql "root" user) Because I did not found the *reinstall* option on sources.. Thanks, Claudio

2 1

Re: [SR-Users] SIP Trunk Usage Monitoring
by JR Richardson 10 May '11

10 May '11

> On 09.05.2011 22:44, JR Richardson wrote: > > Would there be any usage examples of dialog module? I'm not sure if I > > really need profiles and [values] and when to set and unset. A push > > in the right direction would be helpful. > > Hi, > here is an example I've used for incoming and outgoing calls tracking, > just tune it for your purposes: > > loadmodule "dialog.so" > modparam("dialog", "dlg_flag", 4) > modparam("dialog", "db_mode", 1) > modparam("dialog", "profiles_with_value", "carrierin;carrierout") > ... > create_dialog(); > if(is_from_gw()) > { > # Monitor number of incoming calls > get_profile_size("carrierin","$si","$avp(s:cnt)"); > xlog("L_INFO", "----- currently, the carrier $si has > $avp(s:cnt) active outgoing calls\n"); > set_dlg_profile("carrierin","$si"); > } else { > # Monitor number of outgoing calls > get_profile_size("carrierout","$rd","$avp(s:cnt)"); > xlog("L_INFO", "----- currently, the carrier $rd has > $avp(s:cnt) active outgoing calls\n"); > set_dlg_profile("carrierout","$rd"); > } > > -- > Sincerely, > Andrew Pogrebennyk Thanks Andrew, very nice example, this will definitely help. Have a great day. JR

1 0

Re: [SR-Users] Dialog Module Usage Question
by JR Richardson 10 May '11

10 May '11

> > The problem I am having specifically is for outbound calls flowing > > through a dispatcher, I want to separate calls into a profile for each > > SIP domain that I send calls to. I'm not sure how do accomplish this. > > I see in kamctl fifo dlg_list, there is 'caller_bind_addr:: > > udp:10.10.12.24:5060'. Is there a way to export that into a variable > > that I can use as the 'profile' for the dialog? > > The domain part of the request URI is stored in a pseudovariable > called $rd. I would just use that as your profile 'value'. > > -- > Alex Balashov - Principal Thanks Alex, after I walked away from it for a bit, that came to mind. Really appreciate the feedback. Thanks. JR

1 0

Re: [SR-Users] SIP Trunk Usage Monitoring
by JR Richardson 10 May '11

10 May '11

>> I recently interconnected to an upstream carrier using Kamailio 3.0, >> working fine. We have configure 2 SIP trunks for failover/redundnacy. >> I'm using dispatcher module to round robin calls to the carrier. I >> wanted to monitor trunk usage between us. I was reading in the devel >> 3.2 about some native monitoring in the new dispatcher module without >> the need to load the dialog module. >> >> Can I use the new module in version 3.0 or will I need to upgrade the >> core to 3.2? >> >> I typically will not use dev software in production, so can anyone >> suggest a light method of trunk usage monitoring for ver 3.0? I'm not >> opposed to using the dialog module, I'm just not familiar with it and >> not sure the best way to integrate it for usage tracking. I don't >> need to do billing or capture cdr's on the SIP-Routers so the simpler >> the better. > > Honestly, going the dialog module route is your easiest bet. > > -- > Alex Balashov - Principal Thanks Alex, Would there be any usage examples of dialog module? I'm not sure if I really need profiles and [values] and when to set and unset. A push in the right direction would be helpful. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Dialog Module Usage Question
by JR Richardson 10 May '11

10 May '11

Hi All, I'm trying to configure the dialog module with profiles for outbound calls to different Dispatcher domains. I can get the dialog module configured for inbound calls and pull statistics and separate calls into specific profiles, that part is working well. The problem I am having specifically is for outbound calls flowing through a dispatcher, I want to separate calls into a profile for each SIP domain that I send calls to. I'm not sure how do accomplish this. I see in kamctl fifo dlg_list, there is 'caller_bind_addr:: udp:10.10.12.24:5060'. Is there a way to export that into a variable that I can use as the 'profile' for the dialog? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Ser AVP
by Claudio Furrer 09 May '11

09 May '11

Hello all, Have this issue with my ser config. I try to load a fwdbusy value from db using the $fu class, but it's complaining.. config: ... if (avp_db_load("$fu", "s:fwdbusy")) { ... log: 0(20749) ERROR: <core> [pvapi.c:516]: error searching pvar "fu" 0(20749) ERROR: <core> [pvapi.c:720]: wrong char [u/117] in [$fu] at [2 (0)] 0(20749) ERROR: avpops [avpops.c:275]: bad param 1; expected : $pseudo-variable or int/str value What's wrong with that? Should it not considered as a ser avp? Not K one. Or that avp_db_load is an exlusively K function? Thank you, Claudio

2 2

Kamailio fails to register to a VoIP provider
by Pierpaolo Culurciello 09 May '11

09 May '11

Dear all, I'm trying to register Kamailio (version 3.1.3) to different VoIP provider but I always receive a 401 unauthorized even if I use valid credentials. I post below two different capture of the registration to the same VoIP provider using the same credentials (SIP domain 94.23.56.123, SIP auth username 1028), with a Blink softphone (successful) and with Kamailio 3.1.3 (denied). The public IP address of my Kamailio is anonymized with XXXXXXXX: ------ Blink softphone -------- REGISTER sip:94.23.56.123 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:49186;rport;branch=z9hG4bKPjGtybPeuocgJbvOZlRNoWG3RPL0peCs31 Max-Forwards: 70 From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j To: "ItaliADSL" <sip:1028@94.23.56.123> Contact: <sip:alybrkhq@192.168.2.115:49186> Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo CSeq: 1 REGISTER Expires: 600 User-Agent: Blink 0.22.2 (MacOSX) Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.1:49186;branch=z9hG4bKPjGtybPeuocgJbvOZlRNoWG3RPL0peCs31;received=95.248.232.55;rport=49186 From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j To: "ItaliADSL" <sip:1028@94.23.56.123>;tag=as5a82d867 Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo CSeq: 1 REGISTER User-Agent: MOR Softswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6afe07ca" Content-Length: 0 REGISTER sip:94.23.56.123 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:49186;rport;branch=z9hG4bKPj.JA-9Zf0dpKV6v0JmKtOGSMhyNw5ep1w Max-Forwards: 70 From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j To: "ItaliADSL" <sip:1028@94.23.56.123> Contact: <sip:alybrkhq@192.168.2.115:49186> Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo CSeq: 2 REGISTER Expires: 600 User-Agent: Blink 0.22.2 (MacOSX) Authorization: Digest username="1028", realm="asterisk", nonce="6afe07ca", uri="sip:94.23.56.123", response="f3489b3c8510f39bcf3d1fd96b8836fb", algorithm=MD5 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.1:49186;branch=z9hG4bKPj.JA-9Zf0dpKV6v0JmKtOGSMhyNw5ep1w;received=95.248.232.55;rport=49186 From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j To: "ItaliADSL" <sip:1028@94.23.56.123>;tag=as5a82d867 Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo CSeq: 2 REGISTER User-Agent: MOR Softswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 600 Contact: <sip:alybrkhq@192.168.2.115:49186>;expires=600 Date: Sun, 08 May 2011 20:16:58 GMT Content-Length: 0 ----------- Kamailio 3.1.3 ---------------- REGISTER sip:94.23.56.123 SIP/2.0 Via: SIP/2.0/UDP XXXXXXXXXXX;branch=z9hG4bK58b4.3ac3b834.0 To: sip:1028@94.23.56.123 From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-45fd CSeq: 10 REGISTER Call-ID: 6efffe60-27704@XXXXXXXXXXXX Content-Length: 0 User-Agent: kamailio (3.1.3 (i386/linux)) Contact: <sip:1028@XXXXXXXXXX> Expires: 360 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP XXXXXXXXXX;branch=z9hG4bK58b4.3ac3b834.0;received=XXXXXXXXXX From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-45fd To: sip:1028@94.23.56.123;tag=as322a1728 Call-ID: 6efffe60-27704@XXXXXXXXXXX CSeq: 10 REGISTER User-Agent: MOR Softswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f3ee3e8" Content-Length: 0 REGISTER sip:94.23.56.123 SIP/2.0 Via: SIP/2.0/UDP XXXXXXXXXX;branch=z9hG4bK7666.c1b2e93.0 To: sip:1028@94.23.56.123 From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-1f9c CSeq: 10 REGISTER Call-ID: 6efffe60-27697@XXXXXXXXXXX Content-Length: 0 User-Agent: kamailio (3.1.3 (i386/linux)) Contact: <sip:1028@XXXXXXXXXXX> Expires: 360 Authorization: Digest username="1028", realm="asterisk", nonce="5f3ee3e8", uri="sip:94.23.56.123", response="df6ebfa9b5870b8b912d1e3b53eedc62", algorithm=MD5 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP XXXXXXXXX;branch=z9hG4bK7666.c1b2e93.0;received=XXXXXXXXX From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-1f9c To: sip:1028@94.23.56.123;tag=as529fc11d Call-ID: 6efffe60-27697@XXXXXXXXXX CSeq: 10 REGISTER User-Agent: MOR Softswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41f3fe77" Content-Length: 0 -------------------------------------- As you can see from the captures, Blink softphone successfully registrates with VoIP provider while Kamailio's request is rejected. I tried to analize the two flows of SIP messages but I didn't notice any difference between them so I can't understand why Kamailio is not able to register. Can anybody help me to solve this issue? greetings, Pierpaolo

2 1

Authenticate if receiving 401
by Eric Hiller 09 May '11

09 May '11

I want kamailio to authenticate itself to a host if it is sent a 401, just as that host is expected to authenticate if kamailio sends it one. I am not finding much in the online probably because I am not searching for the right terms. Does anyone have any experience in this? Thanks! -Eric

4 4

Kamailio BLF - NOT SIMPLE
by Lucas Alvarez 09 May '11

09 May '11

Hi, I've been trying to configure kamailio for enabling BLF for Aastra phones. Sincerely, it's been a nightmare, till now I'm not sure if I have to enable pua_dialoginfo or presence_dialoginfo module or both for handling dialog events. Besides that is not clear which changes should be made to the route configuration nor which parameters should be configured. I would appreciate if anyone can enlighten me or could point me to the right documentation page. Thanks in advance. Lucas Alvarez

2 1

Force RTPProxy
by Simone Felici 09 May '11

09 May '11

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). Due some network restrictions, we need to force every call to pass trought rtpproxy. Could someone point me to the right place where to force it? Please an example based on our config would be really helpful, but in two scenarios: 1. ALL CALLS trought rtpproxy 2. ONLY CALLS from a given network trought rtpproxy We are in doubt on what choose between the two solutions. Having both examples could me help to implement the needs faster :) Thank's a lot for your help! Simon

2 5

rewriteuri and append_branch with variables
by alex pappas 07 May '11

07 May '11

Dear All, I'm trying the following and i get error: $var(new_uri) = "sip:" +$avp(s:term_prefix) +$rU +"@" +$avp(s:gw_address); rewriteuri=($var(new_uri)); OR append_branch($var(new_uri)); But it seems that it does not like the variable. The documentation says that rewriteuri get a string so i don't understand why does not work. Do you have any ideas on this ? Cheers Alex

3 4

SIP Trunk Usage Monitoring
by JR Richardson 06 May '11

06 May '11

Hi All, I recently interconnected to an upstream carrier using Kamailio 3.0, working fine. We have configure 2 SIP trunks for failover/redundnacy. I'm using dispatcher module to round robin calls to the carrier. I wanted to monitor trunk usage between us. I was reading in the devel 3.2 about some native monitoring in the new dispatcher module without the need to load the dialog module. Can I use the new module in version 3.0 or will I need to upgrade the core to 3.2? I typically will not use dev software in production, so can anyone suggest a light method of trunk usage monitoring for ver 3.0? I'm not opposed to using the dialog module, I'm just not familiar with it and not sure the best way to integrate it for usage tracking. I don't need to do billing or capture cdr's on the SIP-Routers so the simpler the better. Any guidance will be much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

t_check_status documentation
by Eric Hiller 06 May '11

06 May '11

One thing that really threw me this morning was the lack of t_check_status in the modules documentation. Apparently it is still there and works, just completely undocumented. Once I realized it was still there I just went back to the 1.5 docs and was good, but just thought I should point this out. -Eric

2 1

db_text issue
by Eric Hiller 06 May '11

06 May '11

I would like to use dbtext to store a list of whitelisted gateways, but each time I load it I get a core segmentation fault. Any ideas? Here is my gateways db file: key_name(str) key_type(int) value_type(int) key_value(str) gwip:1:0:a gwip:1:0:b gwip:1:0:c gwcount:0:1:3 And in my config: modparam("htable","htable","gateways=>size=4;dbtable=gateways;") modparam("htable","db_url","text:///usr/local/etc/kamailio") while( $var(i) < $sht(gateways=>gwcount) ){ if( $sht(gateways=>gwip[$var(i)]) == $si ){ setflag(gateway); xlog("ISFROMGW: request from IP $si matches $sht(gateways=>gwip[$var(i)]) ...\n"); } } Do you see anything wrong with this setup? -Eric

2 4

free tickets linuxtag
by Henning Westerholt 06 May '11

06 May '11

Hello, we've few free 4 day tickets for the LinuxTag next week in Berlin to share. If you like to receive one, just send me a short reply. More informations about our booth: http://www.kamailio.org/w/2011/04/kamailio-booth-at-linuxtag-2011/ Best regards, Henning

1 0

Force RTPProxy
by Simone Felici 06 May '11

06 May '11

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). Due some network restrictions, we need to force every call to pass trought rtpproxy. Could someone point me to the right place where to force it? Please an example based on our config would be really helpful, but in two scenarios: 1. ALL CALLS trought rtpproxy 2. ONLY CALLS from a given network trought rtpproxy We are in doubt on what choose between the two solutions. Having both examples could me help to implement the needs faster :) Thank's a lot for your help! Simon

1 0

Calling t_on_reply in branch_route doesn't work in Kamailio 3.1 - did work in Kamailio 1.5
by Sebastian Damm 05 May '11

05 May '11

Hi, I'm currently migrating a Kamailio 1.5 installation to Kamailio 3.1.3, and I got all config changes merged, except this one: branch_route[1] { if ((isbflagset(4)) && (!isflagset(7))) { fix_nated_sdp("1"); if (has_body()) { force_rtp_proxy("R"); } else { setflag(5); } t_on_reply("2"); } } In Kamailio 1.5 this worked without problems. In Kamailio 3.1 I get the following error (line 1627 is the line with t_on_reply on it): hostname:~# /usr/sbin/kamailio -c -f /etc/kamailio/proxy/kamailio.cfg loading modules under /urs/lib/kamailio/modules/:/usr/lib/kamailio/modules_k:/usr/lib/kamailio/modules_s/ 0(13670) : <core> [cfg.y:3412]: parse error in config file /etc/kamailio/proxy/kamailio.cfg, line 1627, column 17: Command cannot be used in the block ERROR: bad config file (1 errors) hostname:~# In the Kamailio 1.5 documentation it was stated for every t_on_* function, where it could be called. For t_on_reply it said: This function can be used from REQUEST_ROUTE, BRANCH_ROUTE, ONREPLY_ROUTE and FAILURE_ROUTE. In the Kamailio 3.1 documentation those lines are missing. But I don't really see a reason, why t_on_reply shouldn't be called from branch route. We need it for every branch to possibly have different onreply_routes. Am I missing something? Thanks in advance for any hints. Best regards, Sebastian

3 6

Force RTPProxy
by Simone Felici 05 May '11

05 May '11

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). Due some network restrictions, we need to force every call to pass trought rtpproxy. Could someone point me to the right place where to force it? Please an example based on our config would be really helpful, but in two scenarios: 1. ALL CALLS trought rtpproxy 2. ONLY CALLS from a given network trought rtpproxy We are in doubt on what choose between the two solutions. Having both examples could me help to implement the needs faster :) Thank's a lot for your help! Simon

1 0

Re: [SR-Users] Kamailio lib/srdb1/db_id.c handling of multiple database connections and race condition
by Robert Verspuy 05 May '11

05 May '11

Hi Torge, I think it's better to send your mail to the mailinglist, then more people can have a look at your problem. I'm not really familiar with the kamailio source code. Just started using Kamailio about 8 months back. I came across a problem with requesting information from a postgresql database and made a patch which 'works for me'. I'm not using siptrace. Maybe you can supply more logging (through the mailinglist) of the database communication. The mysql error "Duplicate entry '96193' for key 'PRIMARY'" is a bit weird. As I said, I don't know and don't use the msyql driver in kamailio, but normally I would expect that mysql itself would take care of generating a primary key for the record. So maybe your problem is in the mysql server and not kamailio. Maybe this page can help you: http://www.softwareprojects.com/resources/programming/t-how-to-fix-mysql-du… With kind regards, Robert Verspuy Op 05-05-11 13:19, Dr. Torge Szczepanek schreef: > Hi! > > First of all thanks for all of your work. I really appreciate and I am > glad that there is such a nice Opensource SIP Proxy available. > > In a current project I am heavily relying on SIP Traces stored to a > database and other sqlops, dialplan operations using database connections. > > It seems that there is a race condition with SQL Databases (Postgres > and also MySQL) due to thread insafety as described in: > > http://lists.sip-router.org/pipermail/sr-dev/2010-October/009689.html > > This is causing multiple errors. I guess that the following error is > also caused by this race condition (thread safety): > > /usr/sbin/kamailio[5646]: ERROR: <core> [db_query.c:179]: error while > submitting query > /usr/sbin/kamailio[5646]: ERROR: siptrace [siptrace.c:532]: error > storing trace > /usr/sbin/kamailio[5646]: ERROR: db_mysql [km_dbase.c:120]: driver > error on query: Duplicate entry '96193' for key 'PRIMARY' > > My Kamailo Version (3.1.2) is reporting duplicate database URLs using > a warning: > > /usr/sbin/kamailio[5613]: WARNING: <core> [db_id.c:281]: identical DB > URLs, but different DB connection pid [5613/5562] > /usr/sbin/kamailio[5618]: WARNING: <core> [db_id.c:281]: identical DB > URLs, but different DB connection pid [5618/5562] > /usr/sbin/kamailio[5616]: WARNING: <core> [db_id.c:281]: identical DB > URLs, but different DB connection pid [5616/5562] > /usr/sbin/kamailio[5600]: WARNING: <core> [db_id.c:281]: identical DB > URLs, but different DB connection pid [5600/5562] > /usr/sbin/kamailio[5617]: WARNING: <core> [db_id.c:281]: identical DB > URLs, but different DB connection pid [5617/5562] > > Git commit 9fb37a55e0b97a98b7d2d104a7692f18cb3f7df6 is turning this > into a debug message, which I think is a bad idea if the thread > insafety still exists. > > As far as I know every thread should use its own database handle, at > least when using the MySQL API. > > Is there a way to handle this correctly? Can you give me any clues on > which approach to take here? Since I need a solution for this I would > like to fix this issue at least for myself and of course giving the > change back. > > Greetings Torge > -- > Dr. Torge Szczepanek > > Cygnus Networks GmbH Tel: +49-228-9090987-0 > Adenauerallee 73 Fax: +49-228-9090987-70 > D-53113 Bonn EMail: > info(a)cygnusnetworks.de <mailto:info@cygnusnetworks.de> > Amtsgericht Bonn, HRB 15928 SIP: info(a)cygnusnetworks.de > <mailto:info@cygnusnetworks.de> > Geschaeftsfuehrer Dr. Torge Szczepanek http://www.cygnusnetworks.de/ > Steuernummer 205/5710/1347 USt-ID DE258507007 > -- *Exa-Omicron* Patroonsweg 10 3892 DB Zeewolde Tel.: 088-OMICRON (66 427 66) http://www.exa-omicron.nl

2 1

lcr module: strip and tag
by Alida - Danilo SMALDONE 05 May '11

05 May '11

Hello, I've some understanding problem on lcr module. I've this route route[LCR] { xlog("L_INFO", "ENTRO IN ROUTE LCR ********************************** \n"); if (!load_gws(1)) { sl_send_reply("503", "Unable to load gateways"); exit; } else { $var(i) = 0; while(is_avp_set("$(avp(i:709)[$var(i)])")) { xlog("L_INFO","loading gw_uri_avp[$var(i)]='$(avp(i:709)[$var(i)])'\n"); $var(i) = $var(i) + 1; avp_print(); }; if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "Trying gateway '$avp(i:709)'\n"); } else { xlog("L_INFO", "No More Gateways \n"); }; # try the first matched gateway if (next_gw()) { xlog("L_INFO","ruri_user_avp='$avp(i:500)'\n"); xlog("L_INFO","test ruri_avp_mod='$tu'\n"); #xlog("L_INFO","************************************************************ ****\n"); avp_print(); # Route to failure for failover #t_on_failure("2"); route(RELAY); } else { sl_send_reply("503","No available gateways"); exit; }; }; exit; } And this is what I get: Apr 27 15:44:44 localhost kamailio[7827]: INFO: <script>: ENTRO IN ROUTE LCR ********************************** Apr 27 15:44:44 localhost kamailio[7827]: INFO: <script>: loading gw_uri_avp[0]='1|1|3|774|1171521237|172.16.10.200|5060||1|0' Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1328]: p=0xb60dc5b0, flags=0x0112 Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1334]: id=<709> Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1340]: val_str=<1|1|3|774|1171521237|172.16.10.200|5060||1|0 / 45> Apr 27 15:44:44 localhost kamailio[7827]: INFO: <script>: Trying gateway '1|1|3|774|1171521237|172.16.10.200|5060||1|0' Apr 27 15:44:44 localhost kamailio[7827]: INFO: <script>: ruri_user_avp='09711234567' Apr 27 15:44:44 localhost kamailio[7827]: INFO: <script>: test ruri_avp_mod='sip: 09711234567(a)voip.myproxy.it' Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1328]: p=0xb60dc624, flags=0x0110 Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1334]: id=<712> Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1342]: val_int=<0> Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1328]: p=0xb60dc5b0, flags=0x0112 Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1334]: id=<500> Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1340]: val_str=<09711234567 / 11> I expected that the lcr module strips 3 numbers from avp(i:500) and adds the prefix 774. Anyone can explane why it didn't works? The called number never changes in 77409711891234567. Any help will be appreciated. Thank you Danilo

3 17

Rtp Port Question
by Sungjin Park 04 May '11

04 May '11

Hello, I use your sip service well. But there is one problem occured. I use your sip service on my android system. The application is Csipsimple. On Wifi and 3G connection, your service is very good. your sip account did good work. even 3G network connection. Btw, On Mobile Wimax service, the signal was good. But the Voice could not be trasnsferred. I guess the problem is about rtp port. *I'd like to know the rtp port that IPTEL use*. I will request Korea Telecom about rtp port open. Korea Telecom is Korean mobile Wimax Provider. Thank you. Sungjin Park Seoul, Korea

2 1

Re: [SR-Users] ? about presence
by hb 04 May '11

04 May '11

Thank you for your attention. I use the soft phone "jitsi" as is told in guide file and configure it as the guide. And I have checked the guide twice. I think there have nothing different with the guide excepts the db I used. >From: Martin Pohan?en?k <pohancenik.martin(a)gmail.com> >Subject: Re: [SR-Users] ? about presence >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTi=5S3OHe2SEDDe8ry+X+2ZUd2LqRg(a)mail.gmail.com> >Content-Type: text/plain; charset="gb2312" > >What SIP client app are you using to test presence ? >Please try to post the client configuration, too. > >Also try to go through the entire config file from the guide and compare >what is different from your config file. That might give you a clue of what >is wrong > >2011/5/3 Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> > >> Hello, >> >> If you disable "#!define WITH_PRECENCE" you will not use presence at all - >> so it is most probably that you don't want that. >> When you use db_fallback mode, the subscribers are only written to database >> once in a while (in your case every 20 seconds). >> Also, if the subscribers are not authorized, they will only be in the >> "watchers" table - not active_watchers. >> Can you check if there is something in the "watchers" table? >> Also, can you check your system and see if it works without the xcap >> server? >> >> modparam("presence_xml", "force_active", 1) >> #modparam("presence_xml", "integrated_xcap_server", 1) >> >> Cheers, >> Marius >> >> ------------------------------ >> *From:* hb <hhb.1988(a)163.com> >> *To:* sr-users(a)lists.sip-router.org >> *Sent:* Tue, May 3, 2011 11:44:16 AM >> *Subject:* [SR-Users] ? about presence >> >> Dear everyone: >> I use kamalio 3.1. >> I want to add the xcap_sever and presence modules into my >> kamailio. I followed the lists " >> http://cache.baidu.com/c?m=9d78d513d99a12f00bb6ca255001d7175b14d7226b8b8142…" >> step by step. The only difference is that i use the database postgresql. >> When i start the server, it worked well, but the softphone was not >> able to show the presence of the contacts. >> I tried many times and found that when i disable the sentence >> "#!define WITH_PRECENCE" in "kamailio.cfg", it can show the state of the >> contacts. However, when i check the database, there was nothing in "Active >> Watchers" and "Presentity" tables. >> I think there must have something wrong with the configure of >> presence. Any response will be appreciated. >> >> >> >> ps: the configuration of the presence in kamailio.cfg as follows(same as >> the Tutorial): >> >> #!ifdef WITH_PRESENCE >> loadmodule "presence.so" >> loadmodule "presence_xml.so" >> #!endif >> >> #!ifdef WITH_PRESENCE >> # ----- presence params ----- >> modparam("presence", "db_url", DBURL) >> modparam("presence", "fallback2db", 1) >> modparam("presence", "db_update_period", 20) >> >> # ----- presence_xml params ----- >> modparam("presence_xml", "db_url", DBURL) >> modparam("presence_xml", "force_active", 0) >> modparam("presence_xml", "integrated_xcap_server", 1) >> #!endif >> haibo huang >> >> >> >> ------------------------------ >> ??????2G????????????????????3?!<http://mail.163.com/html/110414_attachment/att1.htm> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20110503/8d425fb…> > >------------------------------ > >Message: 2 >Date: Tue, 3 May 2011 12:55:57 +0900 >From: Sungjin Park <parksj84(a)gmail.com> >Subject: [SR-Users] Rtp Port Question >To: serusers(a)iptel.org >Message-ID: <BANLkTimJwR5e_yM5SA67QjYzOj8s=tgAjQ(a)mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" > >Hello, I use your sip service well. >But there is one problem occured. > >I use your sip service on my android system. >The application is Csipsimple. > >On Wifi and 3G connection, your service is very good. your sip account did >good work. even 3G network connection. >Btw, On Mobile Wimax service, the signal was good. But the Voice could not >be trasnsferred. > > >I guess the problem is about rtp port. >*I'd like to know the rtp port that IPTEL use*. I will request Korea Telecom >about rtp port open. >Korea Telecom is Korean mobile Wimax Provider. > >Thank you. > >Sungjin Park >Seoul, Korea >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20110503/34b8728…> > >------------------------------ > >Message: 3 >Date: Tue, 3 May 2011 12:21:22 -0300 (ART) >From: Claudio Furrer <elcaio(a)gmail.com> >Subject: [SR-Users] INSTALL file >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <alpine.LNX.2.00.1105031219240.29861(a)askepios.local> >Content-Type: TEXT/PLAIN; charset=US-ASCII > >Hello, > >Is this still applicable? > >See INSTALL file (line 669) > >3.1) create MySQL tables for SER flavour > - if you have a previously installed SER on your system, use > /usr/sbin/ser_mysql.sh reinstall > to convert your SER database into new structures > - otherwise, if this is your very first installation, use > /usr/sbin/ser_mysql.sh create > to create SER database structures > (you will be prompted for password of MySql "root" user) > >Because I did not found the *reinstall* option on sources.. > >Thanks, >Claudio > > > >------------------------------ > >Message: 4 >Date: Tue, 3 May 2011 12:49:22 -0300 (ART) >From: Claudio Furrer <elcaio(a)gmail.com> >Subject: [SR-Users] INSTALL file >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <alpine.LNX.2.00.1105031223590.29861(a)askepios.local> >Content-Type: TEXT/PLAIN; charset=US-ASCII > >Hello, > >Is this still applicable? > >See INSTALL file (line 669) > >3.1) create MySQL tables for SER flavour > - if you have a previously installed SER on your system, use > /usr/sbin/ser_mysql.sh reinstall > to convert your SER database into new structures > - otherwise, if this is your very first installation, use > /usr/sbin/ser_mysql.sh create > to create SER database structures > (you will be prompted for password of MySql "root" user) > >Because I did not found the *reinstall* option on sources.. > >Thanks, >Claudio > > > >------------------------------ > >Message: 5 >Date: Wed, 4 May 2011 17:18:17 +0800 (CST) >From: hb <hhb.1988(a)163.com> >Subject: [SR-Users] RE?RE? ? about presence >To: bucur_marius_ovidiu(a)yahoo.com >Cc: sr-users(a)lists.sip-router.org >Message-ID: <694e6d8c.1c32b.12fba4cfaed.Coremail.hhb.1988(a)163.com> >Content-Type: text/plain; charset="gbk" > >Dear?Bucur Marius? > Thank you for your help. > As you said the presence module does not work if I disable it's module. This time I used wireshark to get the packages. I found when I disable the presence module, one point send the "Notify" packages directly to the other, and there were no changes in the watchers table. And when I enable the module, I found it can create some lists in watchers. But at this time , I found there is nothing contents in packages like "Subscribe" and "Notify".What's more when one point change the state, it dose not send any packages to the server. > You said active_watchers will have something when the subscribe is authorised. What dose it mean? My server have not authorised the subscribe? But it can have a call with other subscribe. > Look foreword your help. > haibo huang >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20110504/3f6d4fb…> > >------------------------------ > >_______________________________________________ >sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > >End of sr-users Digest, Vol 72, Issue 5 >***************************************

1 0

RE：RE： ? about presence
by hb 04 May '11

04 May '11

Dear：Bucur Marius： Thank you for your help. As you said the presence module does not work if I disable it's module. This time I used wireshark to get the packages. I found when I disable the presence module, one point send the "Notify" packages directly to the other, and there were no changes in the watchers table. And when I enable the module, I found it can create some lists in watchers. But at this time , I found there is nothing contents in packages like "Subscribe" and "Notify".What's more when one point change the state, it dose not send any packages to the server. You said active_watchers will have something when the subscribe is authorised. What dose it mean? My server have not authorised the subscribe? But it can have a call with other subscribe. Look foreword your help. haibo huang

1 0

INSTALL file
by Claudio Furrer 03 May '11

03 May '11

Hello, Is this still applicable? See INSTALL file (line 669) 3.1) create MySQL tables for SER flavour - if you have a previously installed SER on your system, use /usr/sbin/ser_mysql.sh reinstall to convert your SER database into new structures - otherwise, if this is your very first installation, use /usr/sbin/ser_mysql.sh create to create SER database structures (you will be prompted for password of MySql "root" user) Because I did not found the *reinstall* option on sources.. Thanks, Claudio

1 0

： about presence
by hb 03 May '11

03 May '11

Dear everyone: I use kamalio 3.1. I want to add the xcap_sever and presence modules into my kamailio. I followed the lists "http://cache.baidu.com/c?m=9d78d513d99a12f00bb6ca255001d7175b14d7226b8b8142…" step by step. The only difference is that i use the database postgresql. When i start the server, it worked well, but the softphone was not able to show the presence of the contacts. I tried many times and found that when i disable the sentence "#!define WITH_PRECENCE" in "kamailio.cfg", it can show the state of the contacts. However, when i check the database, there was nothing in "Active Watchers" and "Presentity" tables. I think there must have something wrong with the configure of presence. Any response will be appreciated. ps: the configuration of the presence in kamailio.cfg as follows(same as theTutorial): #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "fallback2db", 1) modparam("presence", "db_update_period", 20) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) #!endif haibo huang

3 2

Error compiling Kamailio with mi_xmlrpc
by Lucas Alvarez 02 May '11

02 May '11

Hi, I'm trying to compile kamailio with mi_xmlrpc, I've followed the instructions, first installing xmlrpc-c-0.9.1.0 and xmlrpc-c-dev-0.9.1.0 but the folder xmlrpc-c in /usr/include is not created, so kamailio couldn't be compiled. Then I've installed the latest version of xmlrpc-c-dev, in this case the folder xmlrpc-c with the headers was created but the module didn;t compile, this are the errors: Makefile:46: You are using an unsupported libxmlrpc-c3 version (1.16.34), compile at your own risk! CC (gcc) [M mi_xmlrpc.so] abyss_data.o abyss_data.c:218: error: expected ')' before '*' token abyss_data.c:235: error: expected ')' before '*' token abyss_data.c:248: error: expected ')' before '*' token abyss_data.c:286: error: expected ')' before '*' token abyss_data.c:298: error: expected ')' before '*' token abyss_data.c:311: error: expected ')' before '*' token abyss_data.c:325: error: expected ')' before '*' token abyss_data.c:331: error: expected ')' before '*' token make: *** [abyss_data.o] Error 1 I would appreciate if someone can point me through the right path. Thanks in advance. Lucas Alvarez

1 0

i want to setup a SIP proxy and i dont know where to start.
by Eliezer Croitoru 02 May '11

02 May '11

i have a main office and a branch office. the main office and the branch office connected via vpn connection the branch office is natted to the main office. so i want to put on the gateway machine a SIP proxy server to nat the sip connection and also to proxy all the connections to SIP uses of the office. means i have one asterisk machine on the main site and other SIP providers that in a case i want to use i can use them also without connecting to the main office. i dont need asterisk on the branch office. the gateway server is ubuntu 10.04 amd64 what i should use Kamailio or the openser on the ubuntu repos? for now the Kamailio is installed.(not working). another feature i want to add is to be able to make calls between SIP phones using this server such as berni(a)mydomain.com from outside the office and vise verse... to dial people at bob(a)otherdomain.com hope for little help Eliezer

2 2

Kamailio performance testing
by Martin Pohančeník 02 May '11

02 May '11

Hi I am trying to test performance of kamailio server. I have made some tests on old hardware, that was ok. I found that it can handle around 450 calls per second with authorization. CPU utilisation was up to 100%. That testing was OK, because server's CPU utilization was increasing. Now I want to do some test with new hardware and do some comparisons of those two hardware configurations. I have made some test but i found that it can handle up to 540 calls per second with authorization. But CPU utilisation was only around 8%. When I increased call rate server responded with 404 Not Here and 500 I am terribly busy messages. I think that it is because of that it is not capable of finding a user location that fast. Can i do something to increase the performacne of this server? I am using SIPp testing tool for generating calls. I have registered users manualy with kamctl ul add sipp sip:sipp@192.168.1.1:5060 command. I was using this tutorial: http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling. And also i made my own scenarios. Any suggestion would be usefull. Thanks.

1 0

New module ipops
by Iñaki Baz Castillo 02 May '11

02 May '11

Hi, I've commited a new module named 'ipops' which provides operations with IPv4 and IPv6: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=5940394… The README describes the module functions: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=modules/… The module has no dependecies on other siprouter modules or external libraries, so if it's ok for you it could be enabled by default. Cheers. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 3

SIP outbound
by Klaus Darilion 02 May '11

02 May '11

Hi! IIRC we recently had a discussion about SIP outbound [1] and the benefits of an implementation in Kamailio. Just playing around with a SIP client on an iphone I discovered a usecase. Whenever the iphone gets a new IP address (quite often as it often changes between WiFi and 3G) the client reREGISTERs with the new IP address (establishing a new TCP connection). This leads to lots of "broken" registrations and TCP connections [2]. By use of SIP outbound and re-registration using the same "reg-id" the proxy would be able to replace the old registration with the new registration. Further, if Kamailio would have a mapping from registrations to TCP connections it could also close the dead TCP connections. regards Klaus [1] http://tools.ietf.org/html/rfc5626 [2] I do not know if TCP keepalives (socket option) could detect broken TCP connections without making too much noise on the receiver side (activating the background application and draining battery)

4 4

Kamailio v3.1.3 in Action
by Andreas Granig 01 May '11

01 May '11

Hi all, If you want to see a full-blown Kamailio v3.1.3 based open-source VoIP system in action, check our latest sip:provider CE v2.2-rc1, which has been released today: http://www.sipwise.com/news/announcements/spce-v2_2rc1-release/ It leverages the capabilities of Kamailio to build a quite complex SIP routing engine using a SIP load-balancer and a SIP proxy/registrar, as well as SEMS acting as b2bua. On top of it, it provides provisioning and billing modules and admin and user web interfaces to form an out-of-the-box VoIP soft-switch. Feedback is very much appreciated. Andreas

1 0

uri_db table version
by Claudio Furrer 29 Apr '11

29 Apr '11

Hello, I've found that scripts/my_create.sql doesn't create the version table anymore (new 3.1 installation). It may be removed since many versions ago, sorry i don't know. Loading the modules/db_mysql and modules_k/uri_db, with a SER flavour compilation (and also a ser config file), it raises: 0(15262) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/srtest 0(15262) DEBUG: db_mysql [km_my_con.c:121]: connection type is Localhost via UNIX socket 0(15262) DEBUG: db_mysql [km_my_con.c:122]: protocol version is 10 0(15262) DEBUG: db_mysql [km_my_con.c:123]: server version is 5.1.51-log 0(15262) ERROR: db_mysql [km_dbase.c:120]: driver error on query: Table 'ser_new.version' doesn't exist 0(15262) ERROR: <core> [db_query.c:101]: error while submitting query 0(15262) ERROR: <core> [db.c:366]: error in db_query Should I create the version table manually or using an old script? Thank you. Regards, Claudio

3 5

Re: [SR-Users] How to get rid of this error message.
by Gary Chen 29 Apr '11

29 Apr '11

Thanks. This fixed error. Is $rc the same as $retcode ? Gary -----Original Message----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Friday, April 29, 2011 1:01 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gary Chen Subject: Re: [SR-Users] How to get rid of this error message. Hello, On 4/29/11 4:32 PM, Gary Chen wrote: > I am using prefix2domain("2") in PDT module like this: > $var(found) = prefix2domain("2"); > and I got this error: > ERROR: <core> [lvalue.c:411]: assignment failed at pos: > (701,56-701,56) Apparently when prefix2domain("2) returns an value to > $var(found), it does not like it. But it still work. > The $var(found) still contains the return value (-1) if no prefix > found. And all my configuration is working. It just generate a lot of > this kind of error in my log file. Am I doing something wrong? How can > I fix it? I will look at it, it might be that the -1 return code is usually associated with an error case. Meanwhile try to replace that line with the following: prefix2domain("2"); $var(found) = $rc; Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

How to get rid of this error message.
by Gary Chen 29 Apr '11

29 Apr '11

I am using prefix2domain("2") in PDT module like this: $var(found) = prefix2domain("2"); and I got this error: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (701,56-701,56) Apparently when prefix2domain("2) returns an value to $var(found), it does not like it. But it still work. The $var(found) still contains the return value (-1) if no prefix found. And all my configuration is working. It just generate a lot of this kind of error in my log file. Am I doing something wrong? How can I fix it? Gary

3 2

Kamailio with RTPoxy
by alex pappas 29 Apr '11

29 Apr '11

Dear all, I have a Kamailio with RTPoxy configured. The purpose is to pass the RTP from this server. When I make calls behind NAT the RTProxy seems to be engaged but in traces I don't see any RTP. Is there anything wrong with the idea or I'm missing something in the configuration ? Thank you Alex

3 5

is $td.did avp a normal avp?
by Juha Heinanen 26 Apr '11

26 Apr '11

domain module sets $td.did at lookup_domain call. $td.did seems to be a normal avp, because i can print its value with xlog by referring to it as $avp(td.did). however, its value is gone at failure route, which makes me suspect that $td.did is not really a normal avp. in route block i have $avp(did) = $td.did; xlog("L_INFO", "at t_relay: <td/td.did> = <$avp(did)/$avp(td.did)>\n"); if (t_relay()) exit; branch route, onreply route, and failure route are set when the above t_relay() takes place. at the end of branch route block, i have xlog("L_INFO", "at end of contact branches: <did/td.did> = <$avp(did)/$avp(td.did)>\n"); and at the beginning of onreply route block: xlog("L_INFO", "at beginning of onreply: <did/td.did> = <$avp(did)/$avp(td.did)>\n"); and at the beginning of failure route block: xlog("L_INFO", "at beginning of contact failure: <did/td.did = <$avp(did)/$avp(td.did)>\n"); when i make a call, i get to syslog: Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13706]: INFO: at t_relay: <td/td.did> = <test.fi/test.fi> Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13706]: INFO: at end of contact branches: <did/td.did> = <test.fi/test.fi> Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13708]: INFO: at beginning of onreply: <did/td.did> = <test.fi/<null>> Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13705]: INFO: at beginning of contact failure: <did/td.did = <test.fi/<null>> why is it that $avp(did) value has been preserved from route/branch route block to onreply/failure blocks, but value of $avp(td.did) is not? i'll cc to jan in case he is not on this list. -- juha

4 12

call for participation: kamailio project at LinuxTag 2011
by Henning Westerholt 26 Apr '11

26 Apr '11

Hello, in about five weeks the kamailio project will (again) be present on the LinuxTag fair, "the most important place for Linux and open source software in Europe" - http://www.linuxtag.org/2011/en.html . If you are in Berlin during this time, plan to visit the LinuxTag or just like to help your favourite SIP server project - please consider supporting us in running our fair booth! Even showing up for one or two hours helps a lot. We need also support to properly present our project, like flyers, a banner, maybe some live demonstrations.. So even if you can't participate in person, there are other ways to help us! Please contact me per mail if you could help at the booth or have some ideas about our project presentation. The actual state of preparations can be found in our wiki: http://sip-router.org/wiki/meetings/berlin_2011 Cheers, Henning

1 1

Bug in LDAP module?
by Pan B. Christensen 26 Apr '11

26 Apr '11

Hello, We are using kamailio 1.5 and we have the following problem with ldap_search: A call comes in. If it is to a VOIP user, it will be routed to kamailio, and eventually go to the ldap lookup route: route[10] to look up the user. This is always successful (LDAP can connect). It will then look up user preferences (call forwarding etc.) from SQL in route[12] and route based on this result in route[13]. If the result is "CFU (call forward unconditional) = on", the URI is changed and route[10] (LDAP lookup) is called again. This is always successful (LDAP can connect). If the result is "CFB (call forward busy) = on" and the user is busy, the URI is changed and route[10] (LDAP lookup) is called again. This is also always successful (LDAP can connect). If the result is "CFNR (call forward no reply) = on", the timeout is changed and the call is delivered to the phone through route[14] (location) and route[15] (relay). If the user does not answer, the URI is changed in failure_route[1] and route[10] is called. For some unknown reason, ldap_search always fails in this scenario with the messages: "ldap_session not found", "couldn't get ldap session" and then "no LDAP entry found". Why? And why does this result in the -1 return code and not the -2 return code for ldap_search? The only differences I can think of is that the unsuccessful scenario comes from FAILURE_ROUTE instead of REQUEST_ROUTE, and that longer time has passed since the original invite. Here's a sample log: NOTE the three ERROR lines, the INFO line and the ROUTE 10 line. Dec 27 13:31:55 [3373]: INVITE received from sip:<CALLER>@<SBC_IP> <source IP: <SBC_IP>> to sip:<CALLEE>@<KAM_IP>: Dec 27 13:31:55 [3373]: ERROR:ldap:ldap_reconnect: [ldapprod]: LDAP reconnect successful Dec 27 13:31:55 [3373]: ROUTE 10: LDAP returned: UID = <UID>, voipSipPrimaryUri = <PRIMARY_URI>, voipExtensionUri = sip:<CALLEE>@<DOMAIN> Dec 27 13:31:55 [3373]: ROUTE 12: Effective user properties for UID <UID>: CFU: off <FORWARDTO>, CFB: off <FORWARDTO>, CFNR: on:25 <FORWARDTO>, CFNC: off <FORWARDTO>, VM: <CALLEE>, busy_level: 1, DBrows: 1 Dec 27 13:31:55 [3373]: -------------------- Calling sip:<CALLEE>@<DOMAIN> for 25 seconds. -------------------- Dec 27 13:31:55 [3373]: ROUTE 15: Sending 105 INVITE from sip:<CALLER>@<SBC_IP> to sip:<CALLEE>@<CALLEE_IP> (via <<null>>) Dec 27 13:31:55 [3383]: Response 100 Trying to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Dec 27 13:31:55 [3385]: Response 180 Ringing to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Dec 27 13:32:20 [3388]: FAILURE_ROUTE 1: Timeout for: 104 INVITE from <CALLER> -> <CALLEE> Dec 27 13:32:20 [3388]: -------------------- Forwarding on no reply to <FORWARDTO> -------------------- Dec 27 13:32:20 [3388]: ERROR:ldap:get_connected_ldap_session: [ldapprod]: ldap_session not found Dec 27 13:32:20 [3388]: ERROR:ldap:lds_search: [ldapprod]: couldn't get ldap session Dec 27 13:32:20 [3388]: INFO:ldap:ldap_search_impl: no LDAP entry found Dec 27 13:32:20 [3388]: ROUTE 11: Number lookup results for <FORWARDTO>: Node: VOIP. Dec 27 13:32:20 [3388]: CONFIG ERROR: VOIP number not found in LDAP. Hanging up. Dec 27 13:32:20 [3373]: Response 487 Request Cancelled to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Here's the ldap config file: [ldapprod] ldap_server_url = "ldap://ldap.<DOMAIN>" ldap_network_timeout = 500 ldap_client_bind_timeout = 500 kamailio config snippets (in case you need them): # LDAP lookup for UID and primary URI route[10] { ... ldap_search("ldap://ldapprod/<DN>?uid,voipSipPrimaryUri,voipExtensionUri?one?(&(objectClass=voipAddress)(|(voipSipUri=sip:$rU@uio.no)(|(voipSipUri=$var(uri))(voipSipUri=$var(e164uri)))))"); if ($retcode < 1) { switch ($retcode) { case -1: # no LDAP entry found route(11); return(1); case -2: # internal error sl_send_reply("500", "Internal server error"); exit; } } ldap_result("uid/$avp(s:uid)"); $avp(s:exten) = null; if (!ldap_result("voipExtensionUri/$avp(s:exten)")) { xlog("L_INFO", "voipExtensionUri not found.\n"); } if (!ldap_result("voipSipPrimaryUri/$avp(s:p-uri)")) { xlog("L_WARN", "voipSipPrimaryUri not found.\n"); #exit here? } if (isflagset(1)) { xlog("L_INFO", "ROUTE 10: LDAP returned: UID = $avp(s:uid), voipSipPrimaryUri = $avp(s:p-uri), voipExtensionUri = $avp(s:exten)\n"); } if $avp(s:exten) != null { $ru=$avp(s:exten); } else { $ru=$avp(s:p-uri); } route(12); } } # Number lookup. Non-VoIP URI's go here. route[11] { ... } # Database lookup for call forwarding. route[12] { ... } # Call forwarding logic route[13] { if (is_method("INVITE")) { $avp(s:uio_timer_avp) = "300"; if ($avp(s:cfu_status) == "on") { # Forward immediately xlog("L_INFO", "-------------------- $rU forwarded unconditionally to $avp(s:cfu_number). --------------------\n"); $rU = $avp(s:cfu_number); route(10); } ... if ( $var(dlg_busy) >= $avp(s:busy_level) ) { if ($avp(s:cfb_status) == "on") { xlog("L_INFO", "-------------------- $rU has $var(dlg_busy) active calls. Treshold $avp(s:busy_level). Forwarding on busy to $avp(s:cfb_number) --------------------\n"); $rU = $avp(s:cfb_number); route(10); } ... else if ($avp(s:cfnr_status) == "on") { # Forward to phone with timeout then to $avp(s:cfnr_number). # t_set_fr($avp(s:cfnr_timeout) + "000"); # Recommended method in 3.1. Function does not exist in 1.5. Using old method (avp). $avp(s:uio_timer_avp) = $avp(s:cfnr_timeout); xlog("L_INFO", "-------------------- Calling $ru for $avp(s:uio_timer_avp) seconds. --------------------\n"); route(14); } ... } failure_route[1] { ... if $avp(s:cfnr_status) == "on" { if (isflagset(1)) { xlog("L_INFO", "FAILURE_ROUTE 1: Timeout for: $mi $rm from $fU -> $rU\n"); } revert_uri(); $rU = $avp(s:cfnr_number); xlog("L_INFO", "-------------------- Forwarding on no reply to $rU --------------------\n"); route(10); } ... } With kind regards, Pan

3 5

Kamailio stops responding after 10 days or so
by Morten Isaksen 26 Apr '11

26 Apr '11

Hi! Kamailio 3.0.3. I have a strange problem with one of our Kamailio servers. This one is used for routing (with carrierroute) and to send presence information (with pua module) Once every 10 day or so I get this error and then Kamailio stops responding to any SIP packets. Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: WARNING: <script>: Failure route - M=INVITE RURI=sip:8615XXXX@178.XX.XX.XX F=sip:861XXXXX@188.120.93.114:1025 T=sip:86155XXXX@sip1.uni-tel.dk IP=178.XX.XX.XX ID=6de881ec07f9c6494ee589cf208da358(a)10.11.87.206 Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: ERROR: carrierroute [cr_func.c:95]: cannot find AVP 'carrier' Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: ERROR: carrierroute [cr_func.c:805]: invalid carrier id -1 Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: ERROR: <script>: cr_next_domain failed Shared memory size is 128M and over halv is free just before the error. The server is in production and does handle debug>1 well, so I do not have much information in the log files. Private memory is the default size. Any ideas what it could be, or how to investegate further? I think my next steps would be to increase the private memory og to increase children=4 to children=8 -- Morten Isaksen

3 7

Command 'sercmd lcr.dump_rules' out of memory.
by Alexandre Abreu 26 Apr '11

26 Apr '11

Hello. When running 'sercmd lcr.reload' (without any errors) and then 'sercmd lcr.dump_rules' I've got the following message: ERROR: ctl [binrpc_run.c:510]: ERROR: binrprc: rpc_send: too many message chunks Then I increased the CTL module params 'binrpc_max_body_size' and ' binrpc_struct_max_body_size'. Re-run the command: ERROR: ctl [binrpc_run.c:908]: ERROR: binrpc: rpc_add: out of memory kamailio -V version: kamailio 3.1.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: compiled on 19:43:43 Apr 18 2011 with gcc 4.1.2 It doesn't matter if I increase the PKG_SIZE (> 8). The LCR tables are very small. lcr_rule - 445 lines lcr_rule_target - 445 lines lcr_gw - 7 lines Alexandre

2 1

Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 26 Apr '11

26 Apr '11

You are right! There are actually 2 issues with pjsip: - UDP6 transport does not support DNS lookups. Thus you must set an outboundproxy with IPv6 address - pjsua API does not handle IPv6 media (SDP) As Qjsimple is based on pjsip's pjsua API QjSimple of course has the same issue. regards klaus Am 21.04.2011 14:05, schrieb Pavel Segeč: > Klaus, > > still the same, > > here is a log when UDP is used, there is only A record resolving > > QjSimple: starting initialization/registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: registration_enabled=TRUE > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > > and here when UDP6 > > QjSimple: starting initialization/registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP6 transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: registration_enabled=TRUE > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > > there is no DNS at all > > pavel > > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion >> Sent: Thursday, April 21, 2011 1:27 PM >> To: sr-users(a)lists.sip-router.org >> Subject: Re: [SR-Users] Ipv6 SIP UA >> >> Pavel, I have updated QjSimple to latest pjsip version (0.8.10). It >> would be great if you can test again. If the problem still exists I will >> post a bug report for pjsip. >> >> http://www.ipcom.at/fileadmin/ipcom/files/qjsimple/QjSimple-0.6.6.zip >> >> regards >> Klaus >> >> Am 21.04.2011 10:10, schrieb Pavel Segeč: >>> Hi Klaus, >>> >>> I'm testing your QJSimple client and I'm observing following problems. >> In a IPv4 environments (win 7/32 or 64bit), the client is using a >> classical DNS resolving with SRV and so on, and I'm able to register. >> But inside of the IPV6 environment (IPv4 disabled), when UDP transport >> is used, the client is making just the A DNS record resolving on my >> domain name, not SRV at all. >>> >>> Here is debug log: >>> >>> QjSimple: starting registration ... >>> QjSimple: initializing SIP stack ... >>> QjSimple: pjsua_create succeeded ... >>> QjSimple: "qjsimple" memory pool created ... >>> QjSimple: DLL loaded ... >>> QjSimple: GetNetworkParams found ... >>> QjSimple: GetNetworkParams done ... >>> QjSimple: cfg.nameserver_count=0 >>> QjSimple: pjsua_init succeeded ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: adding UDP transport ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp >>> QjSimple: publish_enabled=FALSE >>> QjSimple: pjsua_acc_add succeeded ... >>> QjSimple: finished intializing SIP stack, start pjsua ... >>> QjSimple: initializing SIP stack ... done >>> QjSimple: Info: on_reg_state_slot called >>> QjSimple: Info: on_reg_state_slot: status=502 >>> QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway >>> >>> When I setup as the transport protocol UDP6 there is no actions at >> all, no dns, no sip, he re is debug log >>> >>> QjSimple: starting registration ... >>> QjSimple: initializing SIP stack ... >>> QjSimple: pjsua_create succeeded ... >>> QjSimple: "qjsimple" memory pool created ... >>> QjSimple: DLL loaded ... >>> QjSimple: GetNetworkParams found ... >>> QjSimple: GetNetworkParams done ... >>> QjSimple: cfg.nameserver_count=0 >>> QjSimple: pjsua_init succeeded ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: adding UDP6 transport ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp >>> QjSimple: publish_enabled=FALSE >>> QjSimple: pjsua_acc_add succeeded ... >>> QjSimple: finished intializing SIP stack, start pjsua ... >>> QjSimple: initializing SIP stack ... done >>> >>> regards >>> >>> pavel >>> >>>> -----Original Message----- >>>> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >>>> bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion >>>> Sent: Wednesday, April 20, 2011 12:49 PM >>>> To: sr-users(a)lists.sip-router.org >>>> Subject: Re: [SR-Users] Ipv6 SIP UA >>>> >>>> >>>> >>>> Am 20.04.2011 11:01, schrieb Pavel Segeč: >>>>> Hi, >>>>> >>>>> yes I can agree now, linphone is working well, qjsimple has some >>>> problem >>>> >>>> If you find some problems (except that it does not work over TCL/TLS: >>>> http://trac.pjsip.org/repos/ticket/1127) please let me know. >>>> >>>> klaus >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>>> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6057 >>>> (20110420) __________ >>>> >>>> Tuto spravu preveril ESET NOD32 Antivirus. >>>> >>>> http://www.eset.sk >>>> >>>> >>>> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 >>>> (20110420) __________ >>>> >>>> Tuto spravu preveril ESET NOD32 Antivirus. >>>> >>>> http://www.eset.sk >>>> >>> >>> >>> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 >> (20110420) __________ >>> >>> Tuto spravu preveril ESET NOD32 Antivirus. >>> >>> http://www.eset.sk >>> >>> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6060 >> (20110421) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6060 (20110421) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 2

Try backup route
by qmail＠top-consulting.net 22 Apr '11

22 Apr '11

I'm using Kamailio 3.1 I have a custom routing statement called MYROUTE which is placed like this: route { ... route (MYROUTE); route (RELAY);} Inside MYROUTE I do some custom SQL checking that basically selects the least cost provider for the ongoing call. The MYROUTE routine sets the $rU, $rd and $rp variables in case a provider has been found for the dialed number or fails with Sip code 404. All this works perfectly, except in my database I also have an alternate provider that I only wish to use as backup in case the main provider returns a 5XX code. How do I get Kamailio to try the second provider after the first one failed ? So far I've tried setting up a t_on_failure() command & routine that picks up the backup provider from the database and sets $rU, $rd margin-right:0px;margin-bottom:0px;margin-left:0px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;"> 0(11460) DEBUG: [parser/msg_parser.c:167]: get_hdr_field: cseq : 0(11460) DEBUG: [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 0(11460) DEBUG: [parser/msg_parser.c:103]: found end of header 0(11460) DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! 0(11460) DEBUG: [parser/parse_to.c:174]: DEBUG: add_param: tag=4a2aab60 0(11460) DEBUG: [parser/parse_to.c:803]: end of header reached, state=29 0(11460) DEBUG: siputils [checks.c:76]: totag found 0(11460) DEBUG: rr [loose.c:108]: No Route headers found 0(11460) DEBUG: rr [loose.c:910]: There is no Route HF I believe the problem is that t_on_failure() does not keep all the proper data associated with the call and that's why it fails. Maybe t_on_failure isn't the right command to use here after all (?). So to resume. Here's what I wish to achieve: 1 Call comes through2. Goes through MYROUTE where a provider is selected3. Call is attempted through provider found in #24. If call fails , select another provider5. Call is attempted through provider found in #4 Thanks!

2 1

Try backup route
by qmail＠polarismail.com 21 Apr '11

21 Apr '11

I'm using Kamailio 3.1 I have a custom routing statement called MYROUTE which is placed like this: route { ... route (MYROUTE); route (RELAY);} Inside MYROUTE I do some custom SQL checking that basically selects the least cost provider for the ongoing call. The MYROUTE routine sets the $rU, $rd and $rp variables in case a provider has been found for the dialed number or fails with Sip code 404. All this works perfectly, except in my database I also have an alternate provider that I only wish to use as backup in case the main provider returns a 5XX code. How do I get Kamailio to try the second provider after the first one failed ? So far I've tried setting up a t_on_failure() command & routine that picks up the backup provider from the database and sets $rU, $rd margin-right:0px;margin-bottom:0px;margin-left:0px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;"> 0(11460) DEBUG: [parser/msg_parser.c:167]: get_hdr_field: cseq : 0(11460) DEBUG: [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 0(11460) DEBUG: [parser/msg_parser.c:103]: found end of header 0(11460) DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! 0(11460) DEBUG: [parser/parse_to.c:174]: DEBUG: add_param: tag=4a2aab60 0(11460) DEBUG: [parser/parse_to.c:803]: end of header reached, state=29 0(11460) DEBUG: siputils [checks.c:76]: totag found 0(11460) DEBUG: rr [loose.c:108]: No Route headers found 0(11460) DEBUG: rr [loose.c:910]: There is no Route HF I believe the problem is that t_on_failure() does not keep all the proper data associated with the call and that's why it fails. Maybe t_on_failure isn't the right command to use here after all (?). So to resume. Here's what I wish to achieve: 1 Call comes through2. Goes through MYROUTE where a provider is selected3. Call is attempted through provider found in #24. If call fails , select another provider5. Call is attempted through provider found in #4 Thanks!

1 0

BLF using kamailio
by Lucas Alvarez 21 Apr '11

21 Apr '11

Hi, I have implemented kamailio with asterisk, I want to enable BLF using kamailio instead of asterisk. I would appreciate any kind of help, below I'm pasting my configuration. *I'm testing with an Aastra phone, when phone sends the subscription kamailio answers with:* U 192.168.15.22:5060 -> 192.168.15.108:5060 SIP/2.0 489 Bad Event. Via: SIP/2.0/UDP 192.168.15.108;branch=z9hG4bKa54a155ad67706dca. From: "1104" <sip:1104@192.168.15.22:5060>;tag=71cf5c9949. To: <sip:1104@192.168.15.22:5060>;tag=99b712eabe88b2bf7a20409f3dc7ebf1-f71b. Call-ID: f22823b8b4d58a8e. CSeq: 3430 SUBSCRIBE. Allow-Events: presence.winfo, presence, dialog. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. *This is my presence configuration:* #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:192.168.15.22:5060") modparam("presence_dialoginfo", "force_single_dialog", 1) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif

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New developer: Jason Penton
by Daniel-Constantin Mierla 21 Apr '11

21 Apr '11

Hello, I want to announce Jason Penton as a new registered developer, with write access to GIT repository - his username id is 'jason.penton'. Jason already submitted an IMS-related module that Carsten added to his IMS branch in the last days, therefore expect contributions from Jason in the IMS extensions. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 21 Apr '11

21 Apr '11

Pavel, I have updated QjSimple to latest pjsip version (0.8.10). It would be great if you can test again. If the problem still exists I will post a bug report for pjsip. http://www.ipcom.at/fileadmin/ipcom/files/qjsimple/QjSimple-0.6.6.zip regards Klaus Am 21.04.2011 10:10, schrieb Pavel Segeč: > Hi Klaus, > > I'm testing your QJSimple client and I'm observing following problems. In a IPv4 environments (win 7/32 or 64bit), the client is using a classical DNS resolving with SRV and so on, and I'm able to register. But inside of the IPV6 environment (IPv4 disabled), when UDP transport is used, the client is making just the A DNS record resolving on my domain name, not SRV at all. > > Here is debug log: > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > > When I setup as the transport protocol UDP6 there is no actions at all, no dns, no sip, he re is debug log > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP6 transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > > regards > > pavel > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion >> Sent: Wednesday, April 20, 2011 12:49 PM >> To: sr-users(a)lists.sip-router.org >> Subject: Re: [SR-Users] Ipv6 SIP UA >> >> >> >> Am 20.04.2011 11:01, schrieb Pavel Segeč: >>> Hi, >>> >>> yes I can agree now, linphone is working well, qjsimple has some >> problem >> >> If you find some problems (except that it does not work over TCL/TLS: >> http://trac.pjsip.org/repos/ticket/1127) please let me know. >> >> klaus >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6057 >> (20110420) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 >> (20110420) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 (20110420) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > >

2 1

Re: [SR-Users] Ipv6 SIP UA
by Pavel Segeč 21 Apr '11

21 Apr '11

Hi, and the IMS Boghe client can be used for IPv6 call testing too. pavel > -----Original Message----- > From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- > bounces(a)lists.sip-router.org] On Behalf Of Pavel Segeč > Sent: Thursday, April 21, 2011 10:10 AM > To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > Subject: Re: [SR-Users] Ipv6 SIP UA > > Hi Klaus, > > I'm testing your QJSimple client and I'm observing following problems. > In a IPv4 environments (win 7/32 or 64bit), the client is using a > classical DNS resolving with SRV and so on, and I'm able to register. > But inside of the IPV6 environment (IPv4 disabled), when UDP transport > is used, the client is making just the A DNS record resolving on my > domain name, not SRV at all. > > Here is debug log: > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > > When I setup as the transport protocol UDP6 there is no actions at all, > no dns, no sip, he re is debug log > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP6 transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > > regards > > pavel > > > -----Original Message----- > > From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- > > bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion > > Sent: Wednesday, April 20, 2011 12:49 PM > > To: sr-users(a)lists.sip-router.org > > Subject: Re: [SR-Users] Ipv6 SIP UA > > > > > > > > Am 20.04.2011 11:01, schrieb Pavel Segeč: > > > Hi, > > > > > > yes I can agree now, linphone is working well, qjsimple has some > > problem > > > > If you find some problems (except that it does not work over TCL/TLS: > > http://trac.pjsip.org/repos/ticket/1127) please let me know. > > > > klaus > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6057 > > (20110420) __________ > > > > Tuto spravu preveril ESET NOD32 Antivirus. > > > > http://www.eset.sk > > > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 > > (20110420) __________ > > > > Tuto spravu preveril ESET NOD32 Antivirus. > > > > http://www.eset.sk > > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 > (20110420) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 > (20110420) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 (20110420) __________ Tuto spravu preveril ESET NOD32 Antivirus. http://www.eset.sk

1 0

Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 21 Apr '11

21 Apr '11

Am 20.04.2011 11:01, schrieb Pavel Segeč: > Hi, > > yes I can agree now, linphone is working well, qjsimple has some problem If you find some problems (except that it does not work over TCL/TLS: http://trac.pjsip.org/repos/ticket/1127) please let me know. klaus

2 1

xmlrpc requests counted as unsupported_methods
by Juha Heinanen 21 Apr '11

21 Apr '11

looks like core stats counts xmlrpc requests served by route [xmlrpc_requests] as unsupported_methods. is there anything a user can do about it or is it a bug that should be fixed? -- juha

2 4

Broken Via/reply-matching for natping OPTION
by Andreas Granig 20 Apr '11

20 Apr '11

Hi, When kamailio-3.1 sends OPTIONS for natping, the Via branch is wrong, e.g: OPTIONS sip:192.168.51.1:5060 SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:5062;branch=0. Route: <something> Should be branch=z9hG4bK<something> instead of branch=0, right? When a reply comes back, like this... SIP/2.0 100 Trying. Via: SIP/2.0/UDP 127.0.0.1:5062;branch=0;rport=5062. From: sip:pinger@<something> ... then kamailio logs this: INFO: <core> [forward.c:786]: broken reply to forward - no 2nd via The full debug log looks like this: DEBUG: <core> [parser_helpers.c:196]: path for branch: '<sip:lb@127.0.0.1:5060;lr;received=sip:192.168.51.1:5060>' DEBUG: <core> [parser/msg_parser.c:640]: SIP Reply (status): DEBUG: <core> [parser/msg_parser.c:642]: version: <SIP/2.0> DEBUG: <core> [parser/msg_parser.c:644]: status: <100> DEBUG: <core> [parser/msg_parser.c:646]: reason: <Trying> DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <0>; state=6 DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <5062>; state=16 DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via DEBUG: <core> [receive.c:145]: After parse_msg... DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=9 DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [23]; uri=[sip:192.168.51.1:5060] DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body [sip:192.168.51.1:5060#015#012] DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <1> <OPTIONS> DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction DEBUG: tm [t_lookup.c:1150]: DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 DEBUG: <core> [parser/msg_parser.c:103]: found end of header INFO: <core> [forward.c:786]: broken reply to forward - no 2nd via DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Thanks, Andreas

4 6

planning next major release 3.2
by Daniel-Constantin Mierla 20 Apr '11

20 Apr '11

Hello, soon we should sketch the roadmap for the next major release 3.2. There are some private git branches with lot of work on it, so we need to synchronize a bit and decide about the proper time: before or after summer. Several devels will be in Berlin for Linuxtag, but no so much overlapping for a big face to face meeting, therefore might be better to have a IRC chat conference sometime in the near future. Next week will be fine for me, say Wednesaday till Friday, in the afternoon to allow fair time of the day for US timezones. If you have other proposals, speak up. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 3

Re: [SR-Users] Ipv6 SIP UA
by Daniel-Constantin Mierla 20 Apr '11

20 Apr '11

Hello, On 4/19/11 9:58 AM, Pavel Segeč wrote: > Hi, > > may someone help me and recommend a software SIP client with a good IPv6 > support implemented? I tried Xlite, Bria, Linphone, SIP communicator and > Jitsi but with a bad results. Each have some problem within pure IPv6 > environment. the last sipit I attended (last year, though) was not rich at all in IPv6 softphones. In the past there was a patched version of KPhone we used at FhG Fokus for some tests, but I wouldn't call it production ready: http://old.iptel.org/products/kphone/ From what was listed above, you touched most of the active softphones, afaik ekiga,qutecom and twinkle do not have it, not sure about zoiper. Cheers, Daniel

4 3

Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 19 Apr '11

19 Apr '11

I once wrote a SIP client based on pjsip: http://www.ipcom.at/en/telephony/qjsimple/ AFAIK it works will using IPv6 when using UDP. regards Klaus Am 19.04.2011 09:58, schrieb Pavel Segeč: > Hi, > > may someone help me and recommend a software SIP client with a good IPv6 > support implemented? I tried Xlite, Bria, Linphone, SIP communicator and > Jitsi but with a bad results. Each have some problem within pure IPv6 > environment. > > palo > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6053 > (20110418) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

ser flavour accounting
by Claudio Furrer 19 Apr '11

19 Apr '11

Hello, I'm trying to migrate ser-0.9.x to the current sr-3.1.x and right now trying to figure out the acc functions/parameters. I used to use "acc_db_request" from acc module, but it's not working using acc_db and acc_syslog in conjuntion. acc_db has acc_db_request[0,1] but are not callable from the configuration file, and acc_syslog has acc_db_request, but doesn't matter if i load this module, the function isn't found... loading modules under /usr/lib/siprouter/modules:/usr/lib/siprouter/modules_s 0(23455) : <core> [cfg.y:3412]: parse error in config file /tmp/ser.cfg, line 358, column 74: unknown command, missing loadmodule? line 358 is: acc_db_request("480 Offline User", "missed_calls"); I would appreciate if you can guide me through the ACC documentation or changes that it has suffered.. I tried to track the git log.. but many changes since my old ser :( Thank you Claudio pd: FYI i did this modifications to adjust my acc settings: # Accounting module +loadmodule "acc_syslog.so" loadmodule "acc_db.so" # ----------------- setting module-specific parameters --------------- @@ -139,15 +140,13 @@ # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported # modparam("acc", "log_flag", 1 ) -modparam("acc_db", "log_level", 1) +modparam("acc_syslog", "log_level", 1) # MySQL acc params modparam("acc_db", "db_url", "mysql://ser:heslo@localhost/ser") -modparam("acc_db", "db_flag", 2) -modparam("acc_db", "db_missed_flag", 3) +modparam("acc_db", "log_flag", 2) +modparam("acc_db", "log_missed_flag", 3) modparam("acc_db", "log_fmt", "miocfspD") -# Acc time set to localtime (1) -modparam("acc_db", "db_localtime", 1)

2 1

modules_s/uri_db functions
by Claudio Furrer 19 Apr '11

19 Apr '11

Hello, What happened to checks.c file on uri_db module (modules_s)? Seem to be an empty file now, only headers exist. Is it the same to use modules_k/ version? I'm currently at v3.1.. Regards, Claudio

2 1

tm stats pseudo variables?
by Juha Heinanen 19 Apr '11

19 Apr '11

is there any means to access native tm stats from script? i know about $stat pvs for accessing k versions of tm stats, but i want access to native tm stats. -- juha

3 2

RTPProxy and bridging IPv4/IPv6 with parallel forking
by Dan-Cristian Bogos 19 Apr '11

19 Apr '11

Hey Guys, I was recently playing with gateway-ing IPv4-IPv6 and hit the following scenario: * AOR having contacts on both ipv4 and ipv6 and I wanted to do parallel forking. RTPProxy bridging works without any issue on a normal setup, however the problem shows up when needing to make calls toward rtpproxy to return both sides of bridge or only one (ee and ie combinations). Did any of you experiment with this scenario? EG: Call comes from ipv4, you want to send it to both ipv4 and ipv6. The branch route looks the right place to call rtpproxy but when calling unforce_rtp_proxy() on CANCEL, will rtpproxy be aware about which ports are we trying to cancel? One more issue would be with re-invite which must go out with the same ip of rtpproxy as original INVITE. Here we could store the bridge direction in some route parameter but unfortunately adding route params is not possible in branches. So what do u think? DanB PS: The "normal" setup of forking calls only ipv4 or only ipv6 works smooth, so support of ipv6 in kamailio or rtpproxy is not questionable.

4 10

stats in kamailio
by Ricardo Martinez 19 Apr '11

19 Apr '11

Hello. I’m trying to get statistics to use charts in SIREMIS v2.0. I was able to load the “Memory” statistics but I want to know if there is more information to use from Kamailio, right now I have : $stat(used_size) $stat(real_used_size) $stat(max_used_size) $stat(free_size) $stat(location-users) $stat(location-contacts) Where can I find the complete list of statistics to get from kamailio?. Thanks in advance. Ricardo Martinez.-

5 13

Attended transfer in load-balanced multi-server setup
by Pan B. Christensen 18 Apr '11

18 Apr '11

Hello all, I'm having problems with attended transfer in certain scenarios. Let me first try to document the relevant parts of the setup: There are two load-balanced Asterisk gateways connected to the old Nortel system. Currently, PSTN connections are done by Nortel. We then have two load-balanced kamailio servers. The phones are connected to these. There are several other servers not relevant to this question. Blind transfer works in any scenario. I've been told that attended transfer works in any scenario if the original call is SIP to SIP. If the original call is from Nortel/PSTN to SIP, then attended transfer will usually not work. Scenarios: A call comes in from Nortel/PSTN (user A) through Nortel1 to a SIP device (user B). User B pushes the transfer button (puts current call on hold and makes a new call) and dials a Nortel / PSTN number to user C. This call goes out through the nortel2 server. Users B and C talk a short while before user B pushes the Transfer button again to connect users A and C. In this scenario, the REFER is forwarded by kamailio to nortel1, which replies "SIP/2.0 481 Call leg/transaction does not exist." The call to join is of course on the other asterisk (nortel2). A similar scenario is where user B transfers to another SIP user. This call will only exist in one of the kamailio servers, and Asterisk will give the same response to the REFER. A third scenario is where both calls are handled by the same Asterisk server. This scenario works. I'm assuming I'll have to build REFER-handling logic into kamailio, but am unsure of how to proceed. Any suggestions? Thanks in advance. With kind regards, Pan

2 2

Notifying RLS when XML documents are updated
by Peter Dunkley 18 Apr '11

18 Apr '11

Hello, I am using Kamailio RLS and integrated XCAP Server. When I add a contact with my client it updates my XCAP documents but there is no additional SIP messaging between the client and Kamailio. I have been assured (by the client author) that this is the correct behaviour. However, having looked at both the XCAP Server and RLS module code I cannot see anything that will either inform RLS that a document has been updated or a timer to check for that. Have I missed something obvious? Thanks, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

perlvdb error
by Roman Yeryomin 18 Apr '11

18 Apr '11

Hi! I'm trying to use perlvdb module but seems that it's not working. Seems that needed functions are not exported correctly: kamailio[11525]: ERROR: <core> [db.c:79]: module db_perlvdb does not export db_use_table function Looking at perlvdb.c one can see that module exports it's functions as "perlvdb", not "db_perlvdb". But changing "perlvdb" to "db_perlvdb" results in segfault when running: kamailio[31336]: segfault at 1 ip b7365e3a sp bfa82fb0 error 4 in perlvdb.so[b7362000+6000] Do I misunderstand something completely or there is a bug? Anyone actually tried using this module or it just sits there forgotten? Regards, Roman

3 3

Probing mode howto?
by Spinov Evgeniy 18 Apr '11

18 Apr '11

Hello, all. Is there any possibility to set one of dispatcher destinations to probing mode manually? I can set it to active/inactive, however cannot set it to probing via MI. This is really required as, for instance, when asterisk reload takes like 1 minute to reload ( huge DB ), I want to set node to inactive first to do not make call attempts, and after reload set it to probing. Thank you.

1 0

loose_route security
by Eric Hiller 18 Apr '11

18 Apr '11

As I look and play with loose_route functionality it seems that by simply placing a route: proxyip;lr header in my invite I can bypass any and all security otherwise built into the configuration. Is this the way everyone has it? I have been unable to find any configuration examples online that show how to secure/restrict access to loose_route? Ideas or suggestions? Thanks, Eric

9 27

Fwd: [RAI] SIPit 28 summary
by Klaus Darilion 17 Apr '11

17 Apr '11

-------- Original-Nachricht -------- Betreff: [RAI] SIPit 28 summary Datum: Fri, 15 Apr 2011 10:47:08 -0500 Von: Robert Sparks <rjsparks(a)nostrum.com> An: rai(a)ietf.org Also available at <https://www.sipit.net/SIPit28_summary> RjS -------- SIPit 28 was hosted by Digium in Huntsville, Alabama, USA the week of Arpil 11-15, 2010. There were 54 attendees from 19 companies visiting from 10 countries. We had 40 distinct implementations. The roles represented (some implementations act in more than one role) 34 endpoints 6 proxy/registrars/b2bua/sbcs 1 dedicated event server Implementations using each transport for SIP messages: UDP 98% TCP 95% TLS 68% server-auth, 50% mutual-auth SCTP 10% DTLS 0% 68% of the implementations present supported IPv6. There was one RFC4474 Identity implementation present. For DNS we had support for: Full RFC3263 : 53% SRV only : 33% A records only : 15% no DNS support : 0% Support for various items in the endpoints: 76% replaces 38% 5389stun 26% turn 24% 3489stun 24% ice 21% sip/stun multiplexing 21% outbound 21% gruu 18% path 18% join 18% diversion 6% history-info (there were no implementations of 4244bis) 3% service-route 3% sigcomp Support for various items in the proxy/registrars: 50% path 50% outbound 33% sip/stun multiplexing 20% diversion 18% gruu 17% history-info 17% service-route The endpoints and B2BUAs implemented these methods: 100% INVITE, CANCEL, ACK, BYE 94% REGISTER 91% OPTIONS 91% REFER 82% NOTIFY 82% INFO 79% UPDATE 71% SUBSCRIBE 68% PRACK 47% PUBLISH 29% MESSAGE 94% of the implementations sent RTP from the port advertised for reception (symmetric-rtp). One of the implementations required the other party to use symmetric-rtp. 88% of the UAs present both sent RTCP and paid attention to RTCP they received. 56% of the endpoints present supported SRTP using sdes. There was one dtls-srtp implementations at this SIPit. 67% of the endpoints supported multipart/MIME. There were no implementations present with S/MIME support. 48% of the implementations present followed RFC6026 (corrections to the INVITE transaction) 68% followed RFC4320 (corrections to the non-INVITE transaction) There were 5 SIP Event Server implementations (not counting endpoints that support events only for refer). There were 9 SIP Event Client implementations (not counting endpoints that support events only for refer). These event packages were supported: Server Client 2 3 presence 1 0 presence.winfo 1 6 message-summary 2 5 dialog 0 2 reg 2 1 conference 1 0 xcap-diff 0 1 kpml These packages were supported with the eventlist extension: Server Client 1 0 presence Five of the proxies present still rely only on max-forwards. There was one implementation of fork-loop-fix, but no implementations of max-breadth. Multiparty tests (Thanks to Eoin McLeod for contributing notes for several tests) The Forking test was very well attended. We had the majority of the participants at the multiparty table. Many teams took away good new information from watching their implementation react to multiple merges and to Via stacks with transports switching between IPv4 and IPv6. Forcing multiple 200 Oks was more difficult than usual given the network equipment at hand. We've created a set of standing automated tests that will allow endpoints and proxies to test that scenario at their leisure. The TLS multiparty leveraged DNS to reduce configuration churn. We ensured a full mesh of connections between participating elements. There were some issues uncovered with the values used in Route/Record-Route not matching the configured certificates at an element. We brainstormed a configuration for the next event that will efficiently expose whether elements are correctly opening a new connection (rather than reusing an existing connection) when a new request with a resource belonging to an authority not matching what had been authenticated on the existing connection arrived. The SRTP tests showed a high level of interoperability (all with SDES keying), including hold/resume and transfer scenarios. There was a successful test of a call with media running long enough to increment the ROC, followed by a hold/resume. Several participants were trying variants of "best-effort/opportunistic" encryption with lower levels of success. These elements tested providing offers with both RTP/AVP and RTP/SAVP media-lines describing the same logical media channel, and with multiple RTP/AVP media lines decorated with different crypto attributes. The STUN/TURN/ICE tests had very high levels of interoperability. The test scenario involved elements in two natted networks sharing the same address space and elements in the public address space (including a separate TURN server for each of the private network islands). We had a mix of elements that supported relay allocation and elements that presented only host/reflexive candidates. A few elements assumed that all of the m-lines in an offer had to present ICE candidates. SDP with a mix of ICE enabled m-lines and m-lines without candidates caused ICE failures. The Outbound test showed several successful flows being created and maintained. A mesh of edge-proxies and proxy/registrars were set up and endpoints established flows through several combinations. There was good UA support for the Flow-Timer, but there were implementations present that didn't randomize the keepalive value. There was not much testing of recovery from failed flows at this event - we'll focus on that at SIPit 29. We held a long multiparty session focusing on exchanging video with constrained bandwidth or impaired (latent or lossy) networks. There was a high level of basic interoperability and reasonable reaction to mid-call shifts in network conditions, but all parties involved came away with changes they wanted to make in their implementations. The Spiral test presented requests to the participating UAs containing a mix of UDP, TCP, and TLS over IPv4 and IPv6 values in large Via and Record-Route stacks. Correct operation of subsequent in-dialog requests in both directions was tested, discovering a few implementation problems, and some invalid assumptions about how dialog-stateful proxies might need to behave, but most scenarios worked well. We followed the usual Spiral test with a "fuzz-ball" test utilizing DNS SRV to randomly distribute the propagation of messages through all of the proxies present, hopping off to the UAs with a low probability, resulting in an easy-to-configure test that exercised a full-mesh of connections between proxies and presented the UAs with very diverse (and large) Via and Record-Route header field values. We exercised the RFC5393 proxy-doom scenario, again leveraging the SRV load-leveling capabilities to simplify configuration. We had messages popping out to the UAs from each level of the tree, giving them a very large-scale merge test. _______________________________________________ RAI mailing list RAI(a)ietf.org https://www.ietf.org/mailman/listinfo/rai

6 11

Meeting in San Francisco
by Daniel-Constantin Mierla 14 Apr '11

14 Apr '11

Hello, I will spend few days in San Francisco area, during May 6-10. If someone wants to meet and discuss about the project, VoIP & stuff, drop me an email. In case there are many people leaving close to SF city, then I can even try to arrange a group meeting, for a dinner or so, to map faces to names and have good drinks -- this decision will be announced later based on feedback, however suggestions about good meeting places are welcome from anyone. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] Problems with siremis custom charts
by Daniel-Constantin Mierla 14 Apr '11

14 Apr '11

Do not send me message privately upon answers on the mailing list - keep the mailing list always cc-ed otherwise the messages will be ignored. On 4/14/11 3:02 PM, Camila Troncoso wrote: > > Daniel, > > I want a bars graph and i´m using another table, one that I create. > The graph displays well with the default configuration (the one that > the other graph of siremis have) but I don´t want an area graph. I > search around siremis files and find a file named charts-lib.php. I > suppose this is the script that creates the graphs , but in this file > are only two type of graphs describe near line 130 ( area and > line_dot). I don’t know if this is why I can´t change the graph type. > yes, so far these are the ones supported by siremis for building custom graphs. The type, one of them, is taken from the xml config files describing the graphs. The library for making graphs supports other types, as you could see in the usrloc charts, but that is using a different php code base than the other grpahs. If you want to enhance it with other types of graphs, that would be great and if you send a working patch then we can include it in the next release. Cheers, Daniel > > Regards, > > Camila > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com > <mailto:miconda@gmail.com>] > *Sent:* jueves, 14 de abril de 2011 6:16 > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* Camila Troncoso > *Subject:* Re: [SR-Users] Problems with siremis custom charts > > Hello, > > On 4/13/11 10:56 PM, Camila Troncoso wrote: > > Hi, > > I´m having problems with siremis charts. I create a new graph and all > is working fine, but I want to use another type of graph ( the default > graphs of siremis uses type=”area”) and when I change the graph type > to any other thing it shows me the same graph but only the area under > the curve isn´t paint , but the curve doesn´t change. > > Please some help with this issue… > > what type of graph you try to use? have you checked the database > statistics table records, do they have different values for this > graph? If not, check if you write the desired values from the kamailio > config file there. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

How I can turn off 100 trying reply
by Derrick Ding 14 Apr '11

14 Apr '11

Hi, All I wonder how I can make Kamailio not to send "100 Trying" msg when receive INVITE request, by modifying kamailio.cfg or source code. Thanks Derrick

4 10

Problems with siremis custom charts
by Camila Troncoso 14 Apr '11

14 Apr '11

Hi, I´m having problems with siremis charts. I create a new graph and all is working fine, but I want to use another type of graph ( the default graphs of siremis uses type=”area”) and when I change the graph type to any other thing it shows me the same graph but only the area under the curve isn´t paint , but the curve doesn´t change. Please some help with this issue… Regards, Camila T. *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 1

Problems with db_text
by Sascha Daniels 14 Apr '11

14 Apr '11

Hi together, I am trying to make a setup with kamailio and freeswitch, that can run with all supported database types. The basic setup is from this howto: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc With mysql everything is (more or less) fine. Creating db_text databases does not work because /opt/kamailio-3.1/share/kamailio/dbtext/kamailio/uacreg is missing and kamdbctl gives up. Just created an empty file, so kamdbctl can finish its work. The subscribers database is created (works in a more simple setup). Now I am getting this error when starting kamailio: http://pastebin.com/vsiUNdCM Version of table: pua:6 Why does the table version not exist? It was created by "kamdbctl create". I am a bit lost with this debug output. Greets Sascha -- AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied --> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister Montabaur B14998 Bücher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de

1 0

Problems with siremis custom charts
by Camila Troncoso 13 Apr '11

13 Apr '11

Hi, I´m having problems with siremis charts. I create a new graph and all is working fine, but I want to use another type of graph ( the default graphs of siremis uses type=”area”) and when I change the graph type to any other thing it shows me the same graph but only the area under the curve isn´t paint , but the curve doesn´t change. Please some help with this issue… Regards, Camila T. *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

1 0

Registrar Server, location and auth_db table
by Santiago Soares 12 Apr '11

12 Apr '11

Hello, I'm trying to use kamailio 1.5.3 as a registrar server. I have a table on my db that contains all user information, like username, password, expire time and ip address, so kamailio should use the same table to authenticate users AND save their location. The problem is that kamailio complains about table version when starts. It expects a different version number for location table and auth_db table. Is there any way I could use the same table for both functions? I'd appreciate any help! Thank you! -- Santiago Soares Fone: (41) 8488-0537

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Re: [SR-Users] sr-users Digest, Vol 71, Issue 40
by Rick Aldred 12 Apr '11

12 Apr '11

Hi Fedrico, Sorry I missed your reply earlier (thanks by the way). I changed the service priority using the Services GUI utility under System | Administration as I didn't know about the update-rc.d command at the time. Maybe the GUI doesn't do all that's required under the hood or maybe I missed something, but anyway it's working now so that's great. Regards, Rick ------------------------------ Hi, from the log you pasted is clear that the problem is that kamailio is starting before mysql. This because, as you said, they have the same priority and so, being the starting order alphabetical for the same priority, kamailio will start before mysql and cannot connect to mysql which is still not up. You told that changing kamailio priority didn't change the result. How did you change the priority? Did you use the update-rc.d command? Regards, Federico

1 0

OT - Sipwise is hiring SIP and Web hackers
by Andreas Granig 11 Apr '11

11 Apr '11

Hi guys, We're looking for experienced SIP (Kamailio, Sems) and Web (Perl/Catalyst, MySQL, HTML/CSS/jQuery) hackers to work on our sip:provider CE platform and their commercial counter-parts. It's on-site (Vienna/Austria), full-time (40h/week) positions, so please no consultancy or free-lancing offers. http://www.sipwise.com/news/jobs/sipwise-is-hiring-perl-web-developers-arou… http://www.sipwise.com/news/jobs/sipwise-is-hiring-voip-engineers-around-vi… If you're interested, please drop us an email at office(a)sipwise.com. Thanks, Andreas

1 0

pua_xmpp presence problem with notify
by Martin Pohančeník 11 Apr '11

11 Apr '11

Hi ! I have managed to get the XMPP module working for IM. Now I am working on Presence service with the pua_xmpp module. I have experienced some problems and I would like to ask some questions to clarify my understandig of Kamailio. When a SUBSCRIBE arrives to Kamailio, it adds a subscribtion to the watchers table in the database. And if the subscriber is online, he is also in the active_watchers table, right? Next... when a PUBLISH request arrives to Kamailio, by default the handle_publish() function is called. This function generates all NOTIFY msgs (based on the database content of watchers and presence rules) and send them to the recipients. I wanted to ask if these generated NOTIFY msgs go through the routing logic, or not ? As I was saying, I was trying to get pua_xmpp module working, but Kamailio does not seem to send NOTIFY to users in XMPP (winfo is sent, bude presence itself is not). Here is a piece of my configuration file: if(is_method("NOTIFY")) { xlog(" ==== NOTIFY msg ====\n"); } if(is_method("NOTIFY") && uri=~"sip:.+[*].+@sip.sk") { xlog(" ====== NOTIFY from $fu to $tu ======\n"); if(pua_xmpp_notify()) { xlog(" ====== NOTIFY to xmpp domain from $fu to $tu ======\n"); t_reply("200", "OK"); } exit; } if(is_method("PUBLISH")) { xlog("===== PUBLISH FROM $fu ====="); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); if(uri=~"sip:.+[*].+@sip.sk" && $hdr(Event)== "presence") { xlog(" ====== SUBSCRIBE for XMPP domain from $fu to see $tu ======\n"); pua_xmpp_req_winfo("$ruri", "$hdr(Expires)"); } t_release(); } exit; Based on this configuration, I never get the " ==== NOTIFY msg ====" output to the syslog. So I assume the NOTIFY never goes through the routing logic. Is that right?? If so, how can I get it to pass the routing logic so the pua_xmpp_notify() function can be executed ?? Is there any way to do this?? Regards Martin

2 2

Re: [SR-Users] Kamailio and several Asterisk - Realtime DB Integration (solved)
by Alejandro Rios P. 11 Apr '11

11 Apr '11

Hello Daniel Thanks for pointing out to the SIP traces. I found out that it if the request is to be sent to a destination using a port other than 5060, it is required to add the $uac_req(ouri) parameter too: $uac_req(ouri)="sip:" + $var(asterisk_ip) + ":" + $var(asterisk_port); .... $uac_req(ouri)="sip:" + $var(asterisk2_ip) + ":" + $var(asterisk2_port); Thanks! Alejandro Rios Peña <alerios(a)debian.org> DISCLAIMER: The opinions expressed are my own, and not necessarily those of my employer. <http://www.avatar.com.co> 2011/4/8 Daniel-Constantin Mierla <miconda(a)gmail.com> > Hello, > > > On 4/8/11 12:28 AM, Alejandro Rios P. wrote: > >> >> Hi all, >> >> I am testing the scenario described here: >> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb >> >> It works great with one instance of Asterisk, but now that I'm testing >> with two Asterisk instances to load balance, and I see a problem regarding >> the registration forwarding to the second Asterisk. >> >> I tried adding a second "uac_req_send()" bellow the first one, but >> Kamailio only sends the first request and the REGISTER never gets to the >> second Asterisk: >> >> # Forward REGISTER to Asterisk >> route[REGFWD] { >> if(!is_method("REGISTER")) >> { >> return; >> } >> $var(rip) = $sel(cfg_get.asterisk.bindip); >> $uac_req(method)="REGISTER"; >> $uac_req(ruri)="sip:" + $var(rip) + ":" + >> $sel(cfg_get.asterisk.bindport); >> $uac_req(furi)="sip:" + $au+ "@" + $var(rip); >> $uac_req(turi)="sip:" + $au+ "@" + $var(rip); >> $uac_req(hdrs)="Contact:<sip:" + $au+ "@" >> + $sel(cfg_get.kamailio.bindip) >> + ":" + $sel(cfg_get.kamailio.bindport) >> + ">\r\n"; >> if($sel(contact.expires) != $null) >> $uac_req(hdrs)= $uac_req(hdrs) + "Expires:" + >> $sel(contact.expires) + "\r\n"; >> else >> $uac_req(hdrs)= $uac_req(hdrs) + "Expires:" + >> $hdr(Expires) + "\r\n"; >> uac_req_send(); >> >> $uac_req(all) = null; >> >> $var(rip2) = $sel(cfg_get.asterisk2.bindip); >> $uac_req(method)="REGISTER"; >> $uac_req(ruri)="sip:" + $var(rip2) + ":" + >> $sel(cfg_get.asterisk2.bindport); >> $uac_req(furi)="sip:" + $au+ "@" + $var(rip2); >> $uac_req(turi)="sip:" + $au+ "@" + $var(rip2); >> $uac_req(hdrs)="Contact:<sip:" + $au+ "@" >> + $sel(cfg_get.kamailio.bindip) >> + ":" + $sel(cfg_get.kamailio.bindport) >> + ">\r\n"; >> if($sel(contact.expires) != $null) >> $uac_req(hdrs)= $uac_req(hdrs) + "Expires:" + >> $sel(contact.expires) + "\r\n"; >> else >> $uac_req(hdrs)= $uac_req(hdrs) + "Expires:" + >> $hdr(Expires) + "\r\n"; >> uac_req_send(); >> >> } >> >> I also tried appending another branch and sending the second >> uac_req_send() from there, but it is not being sent. >> >> How can I use uac_req_send() several times for the same request or in >> parallel? >> > it should work one after the other. append_branch() does not affect > uac_req_send() at all, this is a separate request built from scratch. > > Do you get any error message in the syslog? Can you watch the traffic on > all interfaces 'ngrep -d any -qt -W byline port 5060' ? > > I see you assign '$uac_req(all) = null;', it should be '$uac_req(all) = > $null;' - but I guess it was a copy&paste thing. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

2 3

Possible bugs in nathelper
by Peter Dunkley 11 Apr '11

11 Apr '11

Hello, I think I have found two bugs in nathelper in Kamailio 3.1.3. Has anyone else encountered these? Bug one ------- nat_uac_test() doesn't seem to work for requests that arrive over UDP. I tried with the latest versions of X-Lite and Jitsi for Windows and observed the same behaviour with both. As far as I could tell the UDP and TCP REGISTER methods were identical except for transport, Call-ID, tags (and so on). At the moment I have worked around this by treating all UDP requests as if they have come from behind a NAT - but this is not ideal. Bug Two ------- nat_uac_test() with flag bit 8 (SDP checks) set fails with an error if the request does not contain SDP. I don't think this is correct behaviour as it seems quite sensible to me to push all requests through the same NAT detection route. In this case it is quite likely that only some of the methods will contain SDP. Surely in this case the correct thing would be for the SDP checks to fail, but not be treated as an error? Thanks for you help, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 5

Setting Global Variables
by Nathaniel L Keeling 11 Apr '11

11 Apr '11

Hello, I am trying to use global variables within my configuration script. I am using Kamailio v3.0.0 and I am following the syntax example written in the sample config file but I keep getting syntax errors. Here is a copy of my syntax and I am using cfg_get.voicemailserver.bindip to get the value. #!ifdef WITH_VOICEMAIL voicemailserver.bindip = "10.0.0.1" desc "Voicemail Server IP Address" voicemailserver.bindport = "5060" desc "Voicemail Server Port" #!endif Here is the error that I am receiving: 0(1875) : <core> [cfg.y:3333]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 17: syntax error 0(1875) : <core> [cfg.y:3333]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 17: unknown config variable 0(1875) : <core> [cfg.y:3330]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 19-28: ERROR: bad config file (3 errors) 0(1875) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy Thanks Nathaniel

2 1

Problems with dispatcher policy in HA/LB kamailio/asterisk environment
by Sören Berger 11 Apr '11

11 Apr '11

Hi all, I currently setup a HA/LB conference. I currently have 1 SER and 2 asterisk conference servers. The goal is to distribute the conferences based on the room number on different server. I use ubuntu 10.04 LTS and the packages from http://deb.kamailio.org/kamailio (kamailio_3.1.2+lucid1_amd64.deb) During the implementation I run into trouble with the "8 - first destination" policy which choose not my second server in case of a failure. The dial in is eg. 330 with round robin to any of my asterisk servers. There they check the room number and append it to the dial number. When somebody calls 330 and enter the room number 12345 the asterisk route it back to kamilio with 33012345 route[ASTERISK_CONFERENCE_AUTH] { if ($rU=~"(33[0-4]$)") { if ( method=="INVITE" ) { ds_select_dst("1","4"); forward(); exit(); } } } When the number 33012345 arrives kamailio decides on which server the conference takes place. For redundancy reasons I use the dispatcher again. If one of the server fails, another will take over the conferences. For the scenario with 2 servers this is trivial. There I also could do it another way. But when I add more conference servers it looks like the best solution to me. Here are my rules for the servers. In this case I used the policy "8" which should use the first destination according to the dispatcher.list file. So if I add more servers I can decide about a order for the fail over servers. route [ASTERISK_CONFERENCE] { if ($rU=~"(33[0-4][0-4][0-9][0-9][0-9][0-9]$)") { if ( method=="INVITE" ) { ds_select_dst("2","8"); forward(); exit(); } } if ($rU=~"(33[0-4][5-9][0-9][0-9][0-9][0-9]$)") { if ( method=="INVITE" ) { ds_select_dst("3","8"); forward(); exit(); } } } distpatcher.list # This group will be used with round robin for getting the room number 1 sip:172.16.1.88:5060 0 1 1 sip:172.16.1.89:5060 0 2 # failover schedule for ast-conf1 2 sip:172.16.1.88:5060 0 1 2 sip:172.16.1.89:5060 0 2 # failover schedule for ast-conf2 3 sip:172.16.1.89:5060 0 1 3 sip:172.16.1.88:5060 0 2 My current problem is, that the policy "8" don't select the other server in case of an failure. I simply blocks and the clients retransmit the request to kamailio.I am currently going crazy with that. :) When I change the policy to round robin (replacing "8" with "4"), everything is working fine. But this an option because in the normal case all calls to a conference room should be placed on the same server. Could this be a bug or something like that? Maybe not the right list here, but I didn't find anything about meetme implementation with more that one server or a HA/LB solution for that and was really wondering about that. Somebody maybe heart from such a implementation? Kind Regards, Sören Berger -- Dipl.-Inf. Sören Berger Rechenzentrum Universität Stuttgart Allmandring 30 70550 Stuttgart soeren.berger(a)rus.uni-stuttgart.de +49-(0)711-685-65871

2 3

New developer: Timo Teräs
by Daniel-Constantin Mierla 11 Apr '11

11 Apr '11

Hello, just to let everyone know that I grnted developer access to Timo, who submitted recently a patch for adding a new module named db_sqlite. As the name suggests, the new module will allot to use SQLite databases as alternative to existing db drivers such as mysql, postgres, a.s.o. Timo will maintain the new module and may help with other modules, as he previously submitted a patch for uac module. His GIT username for commits is: tteras. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 2

is_from_user_enum and SRV records
by Eric Hiller 11 Apr '11

11 Apr '11

We have enum records where 0123456789 points to 0123456789(a)domain.tld. And then multiple SRV records pointing to the actual hosts; _sip._udp SRV record for domain.tld pointing to host1.domain.tld and host2.domain.tld for load balancing. In our config we run is_from_user_enum, which up until adding a second entry for the SRV records worked fine. Now however the is_from_user_enum lookup invariably returns the opposite DNS record for the reply. Ie. host1 is requesting and the dns cache returns host2 and then the is_from_user_enum fails and the reply is dropped. Is there a way to make is_from_user_enum test ALL values of the SRV record? Thanks! -Eric

2 1

Kamailio multihoming
by roman＠dmytriv.com 11 Apr '11

11 Apr '11

Hello everybody!!! J Did any of you know or have configured Kamailio servers in redundant and load distribution mode on geographically dispersed servers. Can you share some ideas how can this be done. I'm thinking about 2 -3 servers that can be used to distribute load and for redundancy as well. Is Kamailio capable of this or this needs to be done on OS level. Thank you Roman

2 3

Re: [SR-Users] Kamailio XCAP for Kamailio IMS
by Daniel-Constantin Mierla 08 Apr '11

08 Apr '11

Hello, can you run kamailio with global parameter 'debug=3' and send the syslog messages for such case? That will give the hints about what is going wrong there. I see some characters are hexa encoded, but better see first the logs. Cheers, Daniel On 4/8/11 11:14 AM, Pavel Segeč wrote: > > Hi, > > I'm using Kamailio SIMPLE AS server as the presence and XCAP solution > as the extension for my Kamailio based IMS platfom. As the base > Presence configuration i'm using the default config as is described > inside of the tutorial from ASIPTO site, it works well for pure ietf > SIP SIMPLE environment. But when using it with the IMS, I sniff during > putting the new record 500 Error. May somebody show me a direction > what should be wrong? > > thanks > > palo > > PUT > /xcap-root/resource-lists/users/palo/index/~~/resource-lists/list%5B@name=%22doubango%22%5D/entry%5B@uri=%22sip:segi@ims1.sip.uniza.sk%22%5D > HTTP/1.1 > > Host: 158.193.139.36:5060 > > Content-Length: 151 > > Connection: Keep-Alive > > User-Agent: XDM-client/OMA1.1 > > X-3GPP-Intended-Identity: palo > > Content-Type: application/xcap-el+xml > > <entry uri="sip:segi@ims1.sip.uniza.sk" > xmlns="urn:ietf:params:xml:ns:resource-lists"> > > <display-name>segi(a)ims1.sip.uniza.sk</display-name> > > </entry>HTTP/1.1 500 Server error > > Via: SIP/2.0/TCP 158.193.152.64:20735 > > Server: kamailio (3.1.2 (i386/linux)) > > Content-Length: 0 > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6025 > (20110408) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

2 2

ring_insert_callid() fixup error
by Juha Heinanen 08 Apr '11

08 Apr '11

i have modparam("siputils", "ring_timeout", 30) but i get Apr 8 15:32:02 sip /usr/sbin/sip-proxy[23748]: ERROR: siputils [ring.c:378]: ring_insert_callid functionality deactivated, you need to set a positive ring_timeout Apr 8 15:32:02 sip /usr/sbin/sip-proxy[23748]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/sip-proxy/sip-proxy.cfg:658 Apr 8 15:32:02 sip /usr/sbin/sip-proxy[23748]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/sip-proxy/sip-proxy.cfg:691 when i start sip router (master version). i looked the code briefly and it tests module param value in fixup function, which i have not seen before. is this a bug or am i missing something? -- juha

2 3

Kamailio and several Asterisk - Realtime DB Integration
by Alejandro Rios P. 08 Apr '11

08 Apr '11

Hi all, I am testing the scenario described here: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb It works great with one instance of Asterisk, but now that I'm testing with two Asterisk instances to load balance, and I see a problem regarding the registration forwarding to the second Asterisk. I tried adding a second "uac_req_send()" bellow the first one, but Kamailio only sends the first request and the REGISTER never gets to the second Asterisk: # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); $uac_req(all) = null; $var(rip2) = $sel(cfg_get.asterisk2.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip2) + ":" + $sel(cfg_get.asterisk2.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip2); $uac_req(turi)="sip:" + $au + "@" + $var(rip2); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } I also tried appending another branch and sending the second uac_req_send() from there, but it is not being sent. How can I use uac_req_send() several times for the same request or in parallel? Thank you, Alejandro Rios Peña DISCLAIMER: The opinions expressed are my own, and not necessarily those of my employer. <http://www.avatar.com.co>

2 1

SIP-XMPP problem, please help
by Martin Pohančeník 08 Apr '11

08 Apr '11

Hello everyone. I am a student at University of Žilina (Faculty of Management Science and Informatics), which has been recently awarded Kamailio 2010 Awards for various articles about SIP and KAMAILIO on portal nil.uniza.sk. I am doing a bachelor thesis on topic SIP/SIMPLE - XMPP traversal, but I am experiencing some problems. I have a working Kamailio server serving domain sip.uniza.sk and a working Jabberd2 server serving domain xmpp.uniza.sk. I am running Kamailio 3.1 with XMPP module with following configuration: fork=yes children=4 log_stderror=yes alias="sip.uniza.sk" ... loadmodule "XMPP.so" ... modparam("xmpp", "backend", "component") modparam("xmpp", "domain_separator", "*") modparam("xmpp", "gateway_domain", "sip.uniza.sk") modparam("xmpp", "xmpp_domain", "xmpp-sip.uniza.sk") modparam("xmpp", "xmpp_host", "server.xmpp.uniza.sk") modparam("xmpp", "xmpp_password", "secret") ... #main routing logic route { ... if( uri=~"sip:.+[*].+@sip.uniza.sk") { # absorb retransmissions ### if (!t_newtran()) { sl_reply_error(); return; } if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } else { t_reply("403","not supported"); exit; } } ... } To my knowledge there is no configuration required on the Jabberd2 server (except the password, that the XMPP component authenticates with). I am trying to enable SIP clients to chat with the XMPP clients and vice versa. SIP clients can send messages to XMPP clients. XMPP modules translates SIP to XMPP and forwards the massage to Jabberd2 server. URIs look like: XMPP client from SIP view: xmpp_client*xmpp.uniza.sk(a)sip.uniza.sk SIP client from XMPP view: sip_client*sip.uniza.sk(a)xmpp-sip.uniza.sk The message arrives to XMPP client, but he can not reply. I have sniffed the traffic on the Kamailio server. The XMPP message arrives to kamailio XMPP module (correct port and everything), but the module somehow ignores it, doesn't translate XMPP to SIP and doesn't send it to the SIP recipient. The sniffed message looks like: <message xmlns='jabber:client' type='chat' to='SIP_client*sip.uniza.sk@ xmpp-sip.uniza.sk' id='mir_47' from='XMPP_client(a)xmpp.uniza.sk/Miranda '><body>hello</body></message> Any suggestions on where can the problem be ?? Do I have to add some lines to routing logic for XMPP module to process the XMPP message (from Jabberd2) for SIP_client ?? Thanks Martin

3 5

TLS certificate on sip-router.org expired
by Timo Reimann 08 Apr '11

08 Apr '11

Heya, just noticed that the TLS certificate on sip-router.org has expired since 9.8.2010. Try an HTTPS connection to the web-page, i.e., https://sip-router.org to see. It seems to be self-signed though. Anyways, I just wanted to let you know. Cheers, --Timo

1 0

Kamailio XCAP for Kamailio IMS
by Pavel Segeč 08 Apr '11

08 Apr '11

Hi, I'm using Kamailio SIMPLE AS server as the presence and XCAP solution as the extension for my Kamailio based IMS platfom. As the base Presence configuration i'm using the default config as is described inside of the tutorial from ASIPTO site, it works well for pure ietf SIP SIMPLE environment. But when using it with the IMS, I sniff during putting the new record 500 Error. May somebody show me a direction what should be wrong? thanks palo PUT /xcap-root/resource-lists/users/palo/index/~~/resource-lists/list%5B@name=%2 2doubango%22%5D/entry%5B@uri=%22sip:segi@ims1.sip.uniza.sk%22%5D HTTP/1.1 Host: 158.193.139.36:5060 Content-Length: 151 Connection: Keep-Alive User-Agent: XDM-client/OMA1.1 X-3GPP-Intended-Identity: palo Content-Type: application/xcap-el+xml <entry uri="sip:segi@ims1.sip.uniza.sk" xmlns="urn:ietf:params:xml:ns:resource-lists"> <display-name>segi(a)ims1.sip.uniza.sk</display-name> </entry>HTTP/1.1 500 Server error Via: SIP/2.0/TCP 158.193.152.64:20735 Server: kamailio (3.1.2 (i386/linux)) Content-Length: 0

1 0

Problems with event_route[htable:mod-init]
by Camila Troncoso 07 Apr '11

07 Apr '11

Hello, I´m using Kamalio 3.1.0 and want to alter some databases (mysql) tables at kamailio starts. I read that the route event_route[htable:mod-init] allows me to do that, but i´m having some problems. I load htable.so module and insert the event_route[] in mi kamailio.conf script but when I start kamailio it returns an error: Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): usrloc Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): registrar Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): uri_db Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): domain Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): permissions Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): group Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): mi_fifo Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): avpops Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): dialog Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): lcr Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): sqlops Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): rtimer Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [sr_module.c:828]: DEBUG: init_mod_child (-127): htable Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: htable [htable.c:185]: rank is (-127) Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: htable [htable.c:191]: executing event_route[htable:mod-init] (1) Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:630]: SIP Request: Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:632]: method: <OPTIONS> Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:634]: uri: <sip:you@kamailio.org> Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bbfa0, 32) called from <core>: parser/msg_parser.c: parse_headers(341) Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bbfa0, 32) returns address 0x834f780 frag. 0x834f768 (size=32) on 1 -th hit Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: <core> [mem/q_malloc.c:366]: qm_malloc(0x82bbfa0, 116) called from <core>: parser/msg_parser.c: get_hdr_field(130) Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: <core> [mem/q_malloc.c:406]: qm_malloc(0x82bbfa0, 116) returns address 0x834f7d0 frag. 0x834f7b8 (size=116) on 1 -th hit Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=2 Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 Apr 7 15:30:22 server4967 /usr/local/sbin/kamailio[31610]: DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via Apr 7 15:30:22 server4967 kamailio: ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe * * * * My event_rout[] is as fallows: event_route[htable:mod-init]{ sql_query("cd", "select prefix from lcr_rule", "rc"); xlog("L_INFO", "dbr = $dbr(rc=>rows)"); if($dbr(rc=>rows)>0){ $var(i) = 0; while($var(i)<$dbr(rc=>rows)){ $var(prefijo)=$dbr(rc=>[$var(i),0]); xlog("L_INFO", "$dbr(rc=>[$var(i),0])"); sql_query("cd", "ALTER TABLE rebalse ADD COLUMN Prefijo_$var(prefijo) INT","rr"); $var(i) = $var(i) + 1; } } sql_result_free("rc"); } I tested the commands inside the event_route[] and they work fine… Can someone help me? Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 1

It's ClueCon Time!
by marketing＠cluecon.com 07 Apr '11

07 Apr '11

1 0

"kamctl fifo debug" does not work
by Klaus Darilion 07 Apr '11

07 Apr '11

root@pbx-kam-1:/home/klaus3000# kamctl fifo debug DEBUG:: 2 root@pbx-kam-1:/home/klaus3000# kamctl fifo debug 0 DEBUG:: 0 Mar 9 11:04:23 pbx-kam-1 /usr/sbin/kamailio[4548]: INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): core.debug has been changed to 0 but: root@pbx-kam-1:/home/klaus3000# kamctl fifo debug DEBUG:: 2 Is this a bug? regards Klaus

2 2

Kamailio at Google Summer of Code 2011
by Daniel-Constantin Mierla 07 Apr '11

07 Apr '11

Hello, for the second year, Kamailio is participating in GSoC program. For 2011, the proposal is to implement a signaling gateway between SIP and JINGLE to enable voice calls between the two networks: http://www.jitsi.org/index.php/GSOC2011/KamailioJingle If you are a student, or you know a student, interested in this project, just write me or to sr-dev(a)lists.sip-router.org. The application starts today and the deadline is April 8, therefore it is quite a short period, you will have to hurry. You can read more about the GSoC program and Kamailio at: http://www.kamailio.org/w/gsoc-2011/ http://www.jitsi.org/index.php/Development/Gsoc2011 Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

7 15

Cannot run Kamailio on public IP
by spamasbl＠gmail.com 07 Apr '11

07 Apr '11

Hello all, I have installed latest Kamailio 3.1.3 on latest ubuntu server. Everything seems to be working well on my LAN. I can successfully register with local Kamailio IP address (192.168.1.137). The issue is that I can not run Kamailio on my public IP: listen=udp:77.XX.XX.XX:5060 listen=udp:192.168.1.137:5060 port=5060 Kamailio runs behind Linksys WRT54GL router with latest Linksys firmware and DMZ is turned on (192.168.1.137). Here is the error I am getting when trying to run kamailio with my public IP: Apr 7 02:57:40 sipbox kamailio[1050]: ERROR: <core> [udp_server.c:400]: ERROR: udp_init: bind(7, 0x82cc6e4, 16) on 77.XX.XX.XX: Cannot assign requested address Do you have any idea what I'm doing wrong ? Thanks !

6 10

off-topic: jobs & projects - figuring out the best solution
by Daniel-Constantin Mierla 06 Apr '11

06 Apr '11

Hello, recently the demand in jobs and projects related to kamailio seemed to increase. I can just say that I am very happy with that. While announcing on the mailing lists is ok, I think we should try to make the process better and more productive, i.e., easier to advertise by publishers and easier to read for those interesting. For that, imo, we need a web page where someone can post such announcements and the others can read it. Since I had this idea in mind for some while, I searched for a dokuwiki plugin that could help, so we can reuse our wiki system. I found bureaucracy plugin, but does not seem to be ready for latest dokuwiki version according to its page: http://www.dokuwiki.org/plugin:bureaucracy But maybe it just works, I should try it instead of waiting, anyhow spare time is not my friend these days... Alternatives would be: - a wordpress plugin - haven't searched for one yet, but there are tons of plugins for WP, maybe one will fit the needs - another web application designed for such purposes My question to you is whether you can recommend a solution which we can evaluate. Ideally (or better said, as many characteristics to be met as possible), the system should be able to: - allow people to post announcements for jobs and projects (with or without prior registration, but with spam prevention - e.g., captcha) - allow to publish details about the job/projects, contact addresses, validity period - allow to invalidate an announcement when the position/project is taken - send an email notification when the announcement is done first time, with a link to the web page of the announcement, so we can send it one to a particular mailing list Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 3

job offer: System administrator internet access services
by Henning Westerholt 06 Apr '11

06 Apr '11

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional IT operation position open in our team in Karlsruhe, Germany at the moment. You would be responsible in a team of motivated system administrators for the operation and maintenance of the radius and CPE auto-provisioning services at 1&1. Furthermore you would support our on-call service for all the services operated from the team, including VoIP. A solid understanding of german is necessary for the job, for more informations (additional requirements, how to apply..) please refer to the opening on the job web page. Of course please feel free to contact me per mail as well. http://www.united-internet.de/JobsJobDetail?__sendingdata=1&JobDetails.jobI… Best regards, Henning Westerholt -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany

1 0

Re: [SR-Users] Kamailio ims / rtpproxy
by Carsten Bock 06 Apr '11

06 Apr '11

Hi, the Kamailio IMS modules are based on the OpenIMSCore-CSCF modules, which have built-in RTPProxy support. I am working on replacing the built-in RTPProxy support in the CSCF modules with the "original" rtpproxy-module of the Kamailio/sip-router.org. It does depend on your config, which RTPProxy support you use.... (but the focus is currently on other parts, be patient). Carsten 2011/4/6 Pavel Segeč <Pavel.Segec(a)fri.uniza.sk>: > Hi, > > I do not understand, may you clarify it? I'm using Kamailio based PCSCF not OIMSC, and kamailio has rtpproxy module...should I'm able to build PCSCF using sip router? > > palo > >> -----Original Message----- >> From: Carsten Bock [mailto:carsten@ng-voice.com] >> Sent: Wednesday, April 06, 2011 11:30 AM >> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users >> Mailing List >> Cc: Pavel Segeč >> Subject: Re: [SR-Users] Kamailio ims / rtpproxy >> >> Hi, >> >> one reason more to replace the built-in RTP-Proxy support of the CSCF >> with the "real" rtpproxy moduie of sip-router.org! ;-) >> >> Carsten >> >> 2011/4/6 Pavel Segeč <Pavel.Segec(a)fri.uniza.sk>: >> > Hi Carsten, >> > >> > Thank you for answer. I did it as you described (anyway, i think, that >> nat traversal is by default on), but there were still some problems. >> After some deeper testing, the PCSCF with RTPproxy works well with two >> Monster client (caller and calle). The problem is with Boghe IMS >> clients, I do not know why, private address inside of the SDP is not >> rewritten, what is interesting, media ports are changing. >> > >> > regards >> > >> > palo >> > >> >> -----Original Message----- >> >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> >> bounces(a)lists.sip-router.org] On Behalf Of Carsten Bock >> >> Sent: Wednesday, April 06, 2011 10:32 AM >> >> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - >> Users >> >> Mailing List >> >> Cc: Pavel Segeč >> >> Subject: Re: [SR-Users] Kamailio ims / rtpproxy >> >> >> >> Hi, >> >> >> >> there is no documentation on the RTPProxy interface of the Kamailio >> >> IMS module yet. >> >> You need to have the following linse in the Kamailio.cfg: >> >> >> >> # Enabled Features on this host: >> >> #!define WITH_NAT >> >> [...] >> >> #!ifdef WITH_NAT >> >> #!define RTPPROXY "udp:127.0.0.1:22222" >> >> #!endif >> >> [...] >> >> #!ifndef WITH_NAT >> >> [...] >> >> #!else >> >> modparam("pcscf","NAT_enable", 1) >> >> modparam("pcscf","ping", 1) >> >> modparam("pcscf","ping_all", 0) >> >> modparam("pcscf","nat_detection_type", 0x17) >> >> modparam("pcscf","rtpproxy_socket", RTPPROXY) >> >> modparam("pcscf","rtpproxy_enable", 1) >> >> modparam("pcscf","rtpproxy_disable_tout", 60) >> >> modparam("pcscf","rtpproxy_retr", 5) >> >> modparam("pcscf","rtpproxy_tout", 1) >> >> #!endif >> >> >> >> Kind regards, >> >> Carsten >> >> >> >> 2011/4/5 Pavel Segeč <pavel.segec(a)fri.uniza.sk>: >> >> > Hi, >> >> > >> >> > >> >> > >> >> > I would like ask you if there is a kamailio ims module >> documentation >> >> > somewhere. I need some deeper information how some things are >> >> implemented, I >> >> > have problem to setup NAT traversal with RTPproxy for our Kamailio >> IMS >> >> > platform. There is no communication among the PCSCF and RTPproxy >> >> server over >> >> > control udp sock. We tried unix socket also with the same results. >> >> > >> >> > >> >> > >> >> > Thanks >> >> > >> >> > >> >> > >> >> > palo73 >> >> > >> >> > http://nil.uniza.sk >> >> > >> >> > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6016 >> >> > (20110405) __________ >> >> > >> >> > Tuto spravu preveril ESET NOD32 Antivirus. >> >> > >> >> > http://www.eset.sk >> >> > >> >> > _______________________________________________ >> >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> >> list >> >> > sr-users(a)lists.sip-router.org >> >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> Carsten Bock >> >> http://www.ng-voice.com >> >> mailto:carsten@ng-voice.com >> >> >> >> _______________________________________________ >> >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> >> sr-users(a)lists.sip-router.org >> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 >> >> (20110405) __________ >> >> >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> >> >> http://www.eset.sk >> >> >> > >> > >> > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 >> (20110405) __________ >> > >> > Tuto spravu preveril ESET NOD32 Antivirus. >> > >> > http://www.eset.sk >> > >> > >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> > sr-users(a)lists.sip-router.org >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> >> >> >> -- >> Carsten Bock >> http://www.ng-voice.com >> mailto:carsten@ng-voice.com >> >> Schomburgstr. 80 >> 22767 Hamburg >> Germany >> >> Mobile +49 179 2021244 >> Office +49 40 34927219 >> Fax +49 40 34927220 >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 >> (20110405) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 (20110405) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com

3 5

Re: [SR-Users] Kamailio ims / rtpproxy
by Alex Balashov 06 Apr '11

06 Apr '11

On 04/06/2011 05:48 AM, Pavel Segeč wrote: > I do not understand, may you clarify it? I'm using Kamailio based > PCSCF not OIMSC, and kamailio has rtpproxy module...should I'm able > to build PCSCF using sip router? Don't worry about this. This is a reference to an ongoing bit of development esoterica related to the fact that Kamailio already has--and has had for a very long time--a module to control rtpproxy. It is not part of the IMS module framework Carsten is developing. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

Re: [SR-Users] Kamailio ims / rtpproxy
by Carsten Bock 06 Apr '11

06 Apr '11

Hi, one reason more to replace the built-in RTP-Proxy support of the CSCF with the "real" rtpproxy moduie of sip-router.org! ;-) Carsten 2011/4/6 Pavel Segeč <Pavel.Segec(a)fri.uniza.sk>: > Hi Carsten, > > Thank you for answer. I did it as you described (anyway, i think, that nat traversal is by default on), but there were still some problems. After some deeper testing, the PCSCF with RTPproxy works well with two Monster client (caller and calle). The problem is with Boghe IMS clients, I do not know why, private address inside of the SDP is not rewritten, what is interesting, media ports are changing. > > regards > > palo > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Carsten Bock >> Sent: Wednesday, April 06, 2011 10:32 AM >> To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users >> Mailing List >> Cc: Pavel Segeč >> Subject: Re: [SR-Users] Kamailio ims / rtpproxy >> >> Hi, >> >> there is no documentation on the RTPProxy interface of the Kamailio >> IMS module yet. >> You need to have the following linse in the Kamailio.cfg: >> >> # Enabled Features on this host: >> #!define WITH_NAT >> [...] >> #!ifdef WITH_NAT >> #!define RTPPROXY "udp:127.0.0.1:22222" >> #!endif >> [...] >> #!ifndef WITH_NAT >> [...] >> #!else >> modparam("pcscf","NAT_enable", 1) >> modparam("pcscf","ping", 1) >> modparam("pcscf","ping_all", 0) >> modparam("pcscf","nat_detection_type", 0x17) >> modparam("pcscf","rtpproxy_socket", RTPPROXY) >> modparam("pcscf","rtpproxy_enable", 1) >> modparam("pcscf","rtpproxy_disable_tout", 60) >> modparam("pcscf","rtpproxy_retr", 5) >> modparam("pcscf","rtpproxy_tout", 1) >> #!endif >> >> Kind regards, >> Carsten >> >> 2011/4/5 Pavel Segeč <pavel.segec(a)fri.uniza.sk>: >> > Hi, >> > >> > >> > >> > I would like ask you if there is a kamailio ims module documentation >> > somewhere. I need some deeper information how some things are >> implemented, I >> > have problem to setup NAT traversal with RTPproxy for our Kamailio IMS >> > platform. There is no communication among the PCSCF and RTPproxy >> server over >> > control udp sock. We tried unix socket also with the same results. >> > >> > >> > >> > Thanks >> > >> > >> > >> > palo73 >> > >> > http://nil.uniza.sk >> > >> > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6016 >> > (20110405) __________ >> > >> > Tuto spravu preveril ESET NOD32 Antivirus. >> > >> > http://www.eset.sk >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> > sr-users(a)lists.sip-router.org >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> >> >> >> -- >> Carsten Bock >> http://www.ng-voice.com >> mailto:carsten@ng-voice.com >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 >> (20110405) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6017 (20110405) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220

2 1

Re: [SR-Users] Kamailio ims / rtpproxy
by Carsten Bock 06 Apr '11

06 Apr '11

Hi, there is no documentation on the RTPProxy interface of the Kamailio IMS module yet. You need to have the following linse in the Kamailio.cfg: # Enabled Features on this host: #!define WITH_NAT [...] #!ifdef WITH_NAT #!define RTPPROXY "udp:127.0.0.1:22222" #!endif [...] #!ifndef WITH_NAT [...] #!else modparam("pcscf","NAT_enable", 1) modparam("pcscf","ping", 1) modparam("pcscf","ping_all", 0) modparam("pcscf","nat_detection_type", 0x17) modparam("pcscf","rtpproxy_socket", RTPPROXY) modparam("pcscf","rtpproxy_enable", 1) modparam("pcscf","rtpproxy_disable_tout", 60) modparam("pcscf","rtpproxy_retr", 5) modparam("pcscf","rtpproxy_tout", 1) #!endif Kind regards, Carsten 2011/4/5 Pavel Segeč <pavel.segec(a)fri.uniza.sk>: > Hi, > > > > I would like ask you if there is a kamailio ims module documentation > somewhere. I need some deeper information how some things are implemented, I > have problem to setup NAT traversal with RTPproxy for our Kamailio IMS > platform. There is no communication among the PCSCF and RTPproxy server over > control udp sock. We tried unix socket also with the same results. > > > > Thanks > > > > palo73 > > http://nil.uniza.sk > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6016 > (20110405) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- Carsten Bock http://www.ng-voice.com mailto:carsten@ng-voice.com

2 1

[PATCH] modules_k/uac: fix from/to restore for small original URI
by Timo Teräs 06 Apr '11

06 Apr '11

Seems that the URI length check is superfluous and fails under certain conditions. It does not make sense for the URI to have zero bytes, so just use the first seen zero byte as end marker. I have a reproducible test case where the restore inserts URI with multiple zero-bytes to wire. This happens if the original URI is smaller than the one we rewrote it to using uac_replace_from. Signed-off-by: Timo Teräs <timo.teras(a)iki.fi> --- modules_k/uac/from.c | 14 ++++++++------ 1 files changed, 8 insertions(+), 6 deletions(-) However, I think the delta encoding used for the RR attribute is flawed. Hostile remote server could rewrite the RR attribute and/or From/To headers in a way to forge it to something it was not in the first place. Additionally the delta-encoded RR attribute breaks if the From/To header isn't exact copy of what we sent. Would it not make more sense to just send the real original header (possibly encrypted) but with a checksum? We could then verify if someone had clobbered the RR attribute and ignore it. And we could always restore the original URI even if the URI we are swapping was modified unexpectedly. diff --git a/modules_k/uac/from.c b/modules_k/uac/from.c index 4657e11..50822b6 100644 --- a/modules_k/uac/from.c +++ b/modules_k/uac/from.c @@ -463,15 +463,17 @@ int restore_from( struct sip_msg *msg, int *is_from ) LM_ERR("new URI shorter than old URI\n"); goto failed; } - for( i=0 ; i<old_uri.len ; i++ ) + for( i=0 ; i<old_uri.len ; i++ ) { new_uri.s[i] ^= old_uri.s[i]; - if (new_uri.len==old_uri.len) { - for( ; new_uri.len && (new_uri.s[new_uri.len-1]==0) ; new_uri.len-- ); - if (new_uri.len==0) { - LM_ERR("new URI got 0 len\n"); - goto failed; + if (new_uri.s[i] == 0) { + new_uri.len = i; + break; } } + if (new_uri.len==0) { + LM_ERR("new URI got 0 len\n"); + goto failed; + } LM_DBG("decoded uris are: new=[%.*s] old=[%.*s]\n", new_uri.len, new_uri.s, old_uri.len, old_uri.s); -- 1.7.1

2 1

Custom voicemail table
by Lucas Alvarez 05 Apr '11

05 Apr '11

Hi, is it possible to change the name of the table voicemessages for voicemail profile in a kamailio-asterisk integration? I mean of the kamailio side. Thanks in advance. Lucas

2 2

Kamailio ims / rtpproxy
by Pavel Segeč 05 Apr '11

05 Apr '11

Hi, I would like ask you if there is a kamailio ims module documentation somewhere. I need some deeper information how some things are implemented, I have problem to setup NAT traversal with RTPproxy for our Kamailio IMS platform. There is no communication among the PCSCF and RTPproxy server over control udp sock. We tried unix socket also with the same results. Thanks palo73 http://nil.uniza.sk

1 0

Kamailio RPMs for v3.1.3
by Daniel-Constantin Mierla 05 Apr '11

05 Apr '11

Hello, the RPM packages for Kamailio recently released v3.1.3 are now available for CentOS, Fedora, RedHat and OpenSUSE, available on the opensuse.org build service: * https://build.opensuse.org/package/show?package=kamailio&project=home%3Akam… <https://build.opensuse.org/package/show?package=kamailio&project=home%3Akam…> Specific download links are: * http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_… * http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_… * http://download.opensuse.org/repositories/home:/kamailio:/telephony/Fedora_… * http://download.opensuse.org/repositories/home:/kamailio:/telephony/openSUS… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Kamailio v3.1.3 Released
by Daniel-Constantin Mierla 05 Apr '11

05 Apr '11

Hello, Kamailio SIP Server v3.1.3 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.2. There is no change to database schema or configuration language structure. Deployments running v3.1.0, v3.1.1 or 3.1.2 are strongly recommended to be upgraded to v3.1.3. For more details about version 3.1.3, visit: http://www.kamailio.org/w/2011/04/kamailio-v3-1-3-released/ Note: existing configuration files for installations on 64b operating systems may need to be updated to specify 'lib64' to the path where the modules are installed (to the value of parameter 'mpath'). See bottom note of the link provided above. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 3

tcp hangup
by Juha Heinanen 05 Apr '11

05 Apr '11

i have two sr proxies talking with each other over tcp. one is running 3.0 and the other 3.1. sometimes i notice that no sip requests or any other tcp packets get through from 3.0 sr host to 3.1 sr host. netstat on 3.0 sr host shows that tcp connection to 3.1 sr host is in ESTABLISHED state whereas netstat on 3.1 sr shows that the same tcp connection is in FIN_WAIT1 state. it takes several minutes before the hanging tcp connection is replaced by a new working one. there is no ip level connectivity problem between the two sr hosts. any suggestions on what might be going on? is there some known bug in 3.0 tcp implementation that could explain this? -- juha

3 12

[PATCH] modules_k/db_sqlite: new sql backend
by Timo Teräs 05 Apr '11

05 Apr '11

Implements the kamailio database API for SQLite. Signed-off-by: Timo Teräs <timo.teras(a)iki.fi> --- modules_k/db_sqlite/Makefile | 15 ++ modules_k/db_sqlite/db_sqlite.c | 90 +++++++ modules_k/db_sqlite/dbase.c | 515 +++++++++++++++++++++++++++++++++++++++ modules_k/db_sqlite/dbase.h | 72 ++++++ 4 files changed, 692 insertions(+), 0 deletions(-) create mode 100644 modules_k/db_sqlite/Makefile create mode 100644 modules_k/db_sqlite/db_sqlite.c create mode 100644 modules_k/db_sqlite/dbase.c create mode 100644 modules_k/db_sqlite/dbase.h diff --git a/modules_k/db_sqlite/Makefile b/modules_k/db_sqlite/Makefile new file mode 100644 index 0000000..9a75593 --- /dev/null +++ b/modules_k/db_sqlite/Makefile @@ -0,0 +1,15 @@ +# $Id$ +# +# WARNING: do not run this directly, it should be run by the master Makefile + +include ../../Makefile.defs +auto_gen= +NAME=db_sqlite.so + +LIBS +=-lsqlite3 + +DEFS+=-DOPENSER_MOD_INTERFACE + +SERLIBPATH=../../lib +SER_LIBS+=$(SERLIBPATH)/srdb1/srdb1 +include ../../Makefile.modules diff --git a/modules_k/db_sqlite/db_sqlite.c b/modules_k/db_sqlite/db_sqlite.c new file mode 100644 index 0000000..8ec16e3 --- /dev/null +++ b/modules_k/db_sqlite/db_sqlite.c @@ -0,0 +1,90 @@ +/* + * $Id$ + * + * SQlite module interface + * + * Copyright (C) 2010 Timo Teräs + * + * This file is part of Kamailio, a free SIP server. + * + * Kamailio is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version + * + * Kamailio is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sys/time.h> +#include <sqlite3.h> + +#include "../../sr_module.h" +#include "../../lib/srdb1/db.h" +#include "dbase.h" + +MODULE_VERSION + +static int sqlite_bind_api(db_func_t *dbb) +{ + if(dbb==NULL) + return -1; + + memset(dbb, 0, sizeof(db_func_t)); + + dbb->use_table = db_sqlite_use_table; + dbb->init = db_sqlite_init; + dbb->close = db_sqlite_close; + dbb->free_result = db_sqlite_free_result; + dbb->query = db_sqlite_query; + dbb->insert = db_sqlite_insert; + dbb->delete = db_sqlite_delete; + dbb->update = db_sqlite_update; + dbb->raw_query = db_sqlite_raw_query; + + return 0; +} + +static cmd_export_t cmds[] = { + {"db_bind_api", (cmd_function)sqlite_bind_api, 0, 0, 0, 0}, + {0, 0, 0, 0, 0, 0} +}; + +static int sqlite_mod_init(void) +{ + sqlite3_initialize(); + + LM_INFO("SQlite library version %s (compiled using %s)\n", + sqlite3_libversion(), + SQLITE_VERSION); + return 0; +} + + +static void sqlite_mod_destroy(void) +{ + LM_INFO("SQlite terminate\n"); + + sqlite3_shutdown(); +} + +struct module_exports exports = { + "db_sqlite", + DEFAULT_DLFLAGS, /* dlopen flags */ + cmds, /* module commands */ + 0, /* module parameters */ + 0, /* exported statistics */ + 0, /* exported MI functions */ + 0, /* exported pseudo-variables */ + 0, /* extra processes */ + sqlite_mod_init, /* module initialization function */ + 0, /* response function*/ + sqlite_mod_destroy, /* destroy function */ + 0 /* per-child init function */ +}; diff --git a/modules_k/db_sqlite/dbase.c b/modules_k/db_sqlite/dbase.c new file mode 100644 index 0000000..b64b327 --- /dev/null +++ b/modules_k/db_sqlite/dbase.c @@ -0,0 +1,515 @@ +/* + * $Id$ + * + * SQlite module core functions + * + * Copyright (C) 2010 Timo Teräs + * + * This file is part of Kamailio, a free SIP server. + * + * Kamailio is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version + * + * Kamailio is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include "../../mem/mem.h" +#include "../../dprint.h" +#include "../../lib/srdb1/db_pool.h" +#include "../../lib/srdb1/db_ut.h" +#include "../../lib/srdb1/db_res.h" +#include "../../lib/srdb1/db_query.h" +#include "dbase.h" + +static time_t sqlite_to_timet(double rT) +{ + return 86400.0*(rT - 2440587.5) + 0.5; +} + +static double timet_to_sqlite(time_t t) +{ + return ((((double) t) - 0.5) / 86400.0) + 2440587.5; +} + +/* + * Initialize database module + * No function should be called before this + */ + +static struct sqlite_connection * db_sqlite_new_connection(const struct db_id* id) +{ + struct sqlite_connection *con; + int rc; + + con = pkg_malloc(sizeof(*con)); + if (!con) { + LM_ERR("failed to allocate driver connection\n"); + return NULL; + } + + memset(con, 0, sizeof(*con)); + con->hdr.ref = 1; + con->hdr.id = (struct db_id*) id; /* set here - freed on error */ + + rc = sqlite3_open_v2(id->database, &con->conn, + SQLITE_OPEN_READWRITE | SQLITE_OPEN_CREATE, NULL); + if (rc != SQLITE_OK) { + pkg_free(con); + LM_ERR("failed to open sqlite database '%s'\n", id->database); + return NULL; + } + + return con; +} + +db1_con_t* db_sqlite_init(const str* _url) +{ + return db_do_init(_url, (void *) db_sqlite_new_connection); +} + + +/* + * Shut down database module + * No function should be called after this + */ + +static void db_sqlite_free_connection(struct sqlite_connection* con) +{ + if (!con) return; + + sqlite3_close(con->conn); + free_db_id(con->hdr.id); + pkg_free(con); +} + +void db_sqlite_close(db1_con_t* _h) +{ + db_do_close(_h, db_sqlite_free_connection); +} + +/* + * Release a result set from memory + */ +int db_sqlite_free_result(db1_con_t* _h, db1_res_t* _r) +{ + if (!_h || !_r) { + LM_ERR("invalid parameter value\n"); + return -1; + } + + if (db_free_result(_r) < 0) + { + LM_ERR("failed to free result structure\n"); + return -1; + } + return 0; +} + +/* + * Store name of table that will be used by + * subsequent database functions + */ +int db_sqlite_use_table(db1_con_t* _h, const str* _t) +{ + return db_use_table(_h, _t); +} + +/* + * Reset query context + */ +static void db_sqlite_cleanup_query(const db1_con_t* _c) +{ + struct sqlite_connection *conn = CON_SQLITE(_c); + int rc; + + if (conn->stmt != NULL) { + rc = sqlite3_finalize(conn->stmt); + if (rc != SQLITE_OK) + LM_ERR("finalize failed: %s\n", + sqlite3_errmsg(conn->conn)); + } + + conn->stmt = NULL; + conn->bindpos = 0; +} + +/* + * Convert value to sql-string as db bind index + */ +static int db_sqlite_val2str(const db1_con_t* _c, const db_val_t* _v, char* _s, int* _len) +{ + struct sqlite_connection *conn; + int ret; + + if (!_c || !_v || !_s || !_len || *_len <= 0) { + LM_ERR("invalid parameter value\n"); + return -1; + } + + conn = CON_SQLITE(_c); + if (conn->bindpos >= DB_SQLITE_MAX_BINDS) { + LM_ERR("too many bindings, recompile with larger DB_SQLITE_MAX_BINDS\n"); + return -2; + } + + conn->bindarg[conn->bindpos] = _v; + ret = snprintf(_s, *_len, "?%u", ++conn->bindpos); + if ((unsigned)ret >= (unsigned) *_len) + return -11; + + *_len = ret; + return 0; +} + +/* + * Send an SQL query to the server + */ +static int db_sqlite_submit_query(const db1_con_t* _h, const str* _s) +{ + struct sqlite_connection *conn = CON_SQLITE(_h); + sqlite3_stmt *stmt; + const db_val_t *val; + int rc, i; + + LM_DBG("submit_query: %.*s\n", _s->len, _s->s); + + rc = sqlite3_prepare_v2(conn->conn, _s->s, _s->len, &stmt, NULL); + if (rc != SQLITE_OK) { + LM_ERR("failed to prepare statement: %s\n", + sqlite3_errmsg(conn->conn)); + return -1; + } + conn->stmt = stmt; + + for (i = 1; i <= conn->bindpos; i++) { + val = conn->bindarg[i-1]; + if (VAL_NULL(val)) { + rc = sqlite3_bind_null(stmt, i); + } else switch (VAL_TYPE(val)) { + case DB1_INT: + rc = sqlite3_bind_int(stmt, i, VAL_INT(val)); + break; + case DB1_BIGINT: + rc = sqlite3_bind_int64(stmt, i, VAL_BIGINT(val)); + break; + case DB1_DOUBLE: + rc = sqlite3_bind_double(stmt, i, VAL_DOUBLE(val)); + break; + case DB1_STRING: + rc = sqlite3_bind_text(stmt, i, + VAL_STRING(val), -1, NULL); + break; + case DB1_STR: + rc = sqlite3_bind_text(stmt, i, + VAL_STR(val).s, VAL_STR(val).len, NULL); + break; + case DB1_DATETIME: + rc = sqlite3_bind_double(stmt, i, timet_to_sqlite(VAL_TIME(val))); + break; + case DB1_BLOB: + rc = sqlite3_bind_blob(stmt, i, + VAL_BLOB(val).s, VAL_BLOB(val).len, + NULL); + break; + case DB1_BITMAP: + rc = sqlite3_bind_int(stmt, i, VAL_BITMAP(val)); + break; + default: + LM_ERR("unknown bind value type %d\n", VAL_TYPE(val)); + return -1; + } + if (rc != SQLITE_OK) { + LM_ERR("Parameter bind failed: %s\n", + sqlite3_errmsg(conn->conn)); + return -1; + } + } + + return 0; +} + +static int decltype_to_dbtype(const char *decltype) +{ + /* SQlite3 has dynamic typing. It does not store the actual + * exact type, instead it uses 'affinity' depending on the + * value. We have to go through the declaration types to see + * what to return. */ + if (strstr(decltype, "INT") != NULL || + strncasecmp(decltype, "SERIAL", 6) == 0) + return DB1_INT; + if (strstr(decltype, "CHAR") != NULL) + return DB1_STRING; + if (strstr(decltype, "TEXT") != NULL) + return DB1_STR; + if (strstr(decltype, "REAL") != NULL || + strstr(decltype, "FLOA") != NULL || + strstr(decltype, "DOUB") != NULL) + return DB1_DOUBLE; + if (strstr(decltype, "BLOB") != NULL) + return DB1_BLOB; + if (strncasecmp(decltype, "TIME", 4) == 0 || + strncasecmp(decltype, "DATE", 4) == 0) + return DB1_DATETIME; + + LM_ERR("sqlite decltype '%s' not recognized, defaulting to int", + decltype); + return DB1_INT; +} + +static str* str_dup(const char *_s) +{ + str *s; + int len = strlen(_s); + + s = (str*) pkg_malloc(sizeof(str)+len+1); + s->len = len; + s->s = ((char*)s) + sizeof(str); + memcpy(s->s, _s, len); + s->s[len] = '\0'; + + return s; +} + +static void str_assign(str* s, const char *_s, int len) +{ + s->s = (char *) pkg_malloc(len + 1); + s->len = len; + memcpy(s->s, _s, len); + s->s[len] = 0; +} + +/* + * Read database answer and fill the structure + */ +int db_sqlite_store_result(const db1_con_t* _h, db1_res_t** _r) +{ + struct sqlite_connection *conn = CON_SQLITE(_h); + db1_res_t *res; + int i, rc, num_rows = 0, num_alloc = 0; + db_row_t *rows = NULL, *row; + db_val_t *val; + + res = db_new_result(); + if (res == NULL) + goto no_mem; + + while (1) { + rc = sqlite3_step(conn->stmt); + if (rc == SQLITE_DONE) { + *_r = res; + return 0; + } + if (rc != SQLITE_ROW) { + LM_INFO("sqlite3_step failed: %s\n", sqlite3_errmsg(conn->conn)); + goto err; + } + if (num_rows == 0) { + /* get column types */ + rc = sqlite3_column_count(conn->stmt); + if (db_allocate_columns(res, rc) != 0) + goto err; + RES_COL_N(res) = rc; + + for (i = 0; i < RES_COL_N(res); i++) { + RES_NAMES(res)[i] = str_dup(sqlite3_column_name(conn->stmt, i)); + RES_TYPES(res)[i] = decltype_to_dbtype(sqlite3_column_decltype(conn->stmt, i)); + } + } + if (num_rows >= num_alloc) { + if (num_alloc) + num_alloc *= 2; + else + num_alloc = 8; + rows = pkg_realloc(rows, sizeof(db_row_t) * num_alloc); + if (rows == NULL) + goto no_mem; + RES_ROWS(res) = rows; + } + + row = &RES_ROWS(res)[num_rows]; + num_rows++; + RES_ROW_N(res) = num_rows; /* rows in this result set */ + RES_NUM_ROWS(res) = num_rows; /* rows in total */ + + if (db_allocate_row(res, row) != 0) + goto no_mem; + + for (i = 0, val = ROW_VALUES(row); i < RES_COL_N(res); i++, val++) { + VAL_TYPE(val) = RES_TYPES(res)[i]; + VAL_NULL(val) = 0; + VAL_FREE(val) = 0; + if (sqlite3_column_type(conn->stmt, i) == SQLITE_NULL) { + VAL_NULL(val) = 1; + } else switch (VAL_TYPE(val)) { + case DB1_INT: + VAL_INT(val) = sqlite3_column_int(conn->stmt, i); + break; + case DB1_BIGINT: + VAL_BIGINT(val) = sqlite3_column_int64(conn->stmt, i); + break; + case DB1_STRING: + /* first field of struct str* is the char* so we can just + * do whatever DB1_STR case does */ + case DB1_STR: + str_assign(&VAL_STR(val), + (const char*) sqlite3_column_text(conn->stmt, i), + sqlite3_column_bytes(conn->stmt, i)); + VAL_FREE(val) = 1; + break; + case DB1_DOUBLE: + VAL_DOUBLE(val) = sqlite3_column_double(conn->stmt, i); + break; + case DB1_DATETIME: + VAL_TIME(val) = sqlite_to_timet(sqlite3_column_double(conn->stmt, i)); + break; + case DB1_BLOB: + str_assign(&VAL_BLOB(val), + (const char*) sqlite3_column_blob(conn->stmt, i), + sqlite3_column_bytes(conn->stmt, i)); + VAL_FREE(val) = 1; + break; + default: + LM_ERR("unhandled db-type\n"); + goto err; + } + } + } + +no_mem: + LM_ERR("no private memory left\n"); +err: + if (res) + db_free_result(res); + return -1; +} + +/* + * Query table for specified rows + * _h: structure representing database connection + * _k: key names + * _op: operators + * _v: values of the keys that must match + * _c: column names to return + * _n: number of key=values pairs to compare + * _nc: number of columns to return + * _o: order by the specified column + */ +int db_sqlite_query(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _op, + const db_val_t* _v, const db_key_t* _c, int _n, int _nc, + const db_key_t _o, db1_res_t** _r) +{ + int rc; + + rc = db_do_query(_h, _k, _op, _v, _c, _n, _nc, _o, _r, + db_sqlite_val2str, + db_sqlite_submit_query, + db_sqlite_store_result); + db_sqlite_cleanup_query(_h); + + return rc; +} + +static int db_sqlite_commit(const db1_con_t* _h) +{ + struct sqlite_connection *conn = CON_SQLITE(_h); + int rc; + + rc = sqlite3_step(conn->stmt); + if (rc != SQLITE_DONE && rc != SQLITE_OK) { + LM_ERR("sqlite commit failed: %s\n", + sqlite3_errmsg(conn->conn)); + return -1; + } + + return 0; +} + +/* + * Insert a row into specified table + * _h: structure representing database connection + * _k: key names + * _v: values of the keys + * _n: number of key=value pairs + */ +int db_sqlite_insert(const db1_con_t* _h, const db_key_t* _k, const db_val_t* _v, + int _n) +{ + int rc = -1; + + rc = db_do_insert(_h, _k, _v, _n, + db_sqlite_val2str, + db_sqlite_submit_query); + if (rc == 0) + rc = db_sqlite_commit(_h); + db_sqlite_cleanup_query(_h); + + return rc; +} + + +/* + * Delete a row from the specified table + * _h: structure representing database connection + * _k: key names + * _o: operators + * _v: values of the keys that must match + * _n: number of key=value pairs + */ +int db_sqlite_delete(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _o, + const db_val_t* _v, int _n) +{ + int rc; + + rc = db_do_delete(_h, _k, _o, _v, _n, + db_sqlite_val2str, + db_sqlite_submit_query); + if (rc == 0) + rc = db_sqlite_commit(_h); + db_sqlite_cleanup_query(_h); + + return rc; +} + + +/* + * Update some rows in the specified table + * _h: structure representing database connection + * _k: key names + * _o: operators + * _v: values of the keys that must match + * _uk: updated columns + * _uv: updated values of the columns + * _n: number of key=value pairs + * _un: number of columns to update + */ +int db_sqlite_update(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _o, + const db_val_t* _v, const db_key_t* _uk, const db_val_t* _uv, + int _n, int _un) +{ + int rc; + + rc = db_do_update(_h, _k, _o, _v, _uk, _uv, _n, _un, + db_sqlite_val2str, + db_sqlite_submit_query); + if (rc == 0) + rc = db_sqlite_commit(_h); + db_sqlite_cleanup_query(_h); + + return rc; +} + +int db_sqlite_raw_query(const db1_con_t* _h, const str* _s, db1_res_t** _r) +{ + return db_do_raw_query(_h, _s, _r, + db_sqlite_submit_query, + db_sqlite_store_result); +} diff --git a/modules_k/db_sqlite/dbase.h b/modules_k/db_sqlite/dbase.h new file mode 100644 index 0000000..de006bf --- /dev/null +++ b/modules_k/db_sqlite/dbase.h @@ -0,0 +1,72 @@ +/* + * $Id$ + * + * SQlite module core functions + * + * Copyright (C) 2010 Timo Teräs + * + * This file is part of Kamailio, a free SIP server. + * + * Kamailio is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version + * + * Kamailio is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +#ifndef DBASE_H +#define DBASE_H + +#include <sqlite3.h> + +#include "../../lib/srdb1/db_pool.h" +#include "../../lib/srdb1/db_con.h" +#include "../../lib/srdb1/db_res.h" +#include "../../lib/srdb1/db_key.h" +#include "../../lib/srdb1/db_op.h" +#include "../../lib/srdb1/db_val.h" + +#define DB_SQLITE_MAX_BINDS 64 + +struct sqlite_connection { + struct pool_con hdr; + + sqlite3 *conn; + int bindpos; + + sqlite3_stmt *stmt; + const db_val_t *bindarg[DB_SQLITE_MAX_BINDS]; +}; + +#define CON_SQLITE(db_con) ((struct sqlite_connection *) db_con->tail) + +db1_con_t* db_sqlite_init(const str* _sqlurl); +void db_sqlite_close(db1_con_t* _h); + +int db_sqlite_free_result(db1_con_t* _h, db1_res_t* _r); + +int db_sqlite_query(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _op, + const db_val_t* _v, const db_key_t* _c, int _n, int _nc, + const db_key_t _o, db1_res_t** _r); +int db_sqlite_insert(const db1_con_t* _h, const db_key_t* _k, const db_val_t* _v, + int _n); +int db_sqlite_delete(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _o, + const db_val_t* _v, int _n); +int db_sqlite_update(const db1_con_t* _h, const db_key_t* _k, const db_op_t* _o, + const db_val_t* _v, const db_key_t* _uk, const db_val_t* _uv, + int _n, int _un); +int db_sqlite_raw_query(const db1_con_t* _h, const str* _s, db1_res_t** _r); + +int db_sqlite_use_table(db1_con_t* _h, const str* _t); + + +#endif /* DBASE_H */ -- 1.7.1

3 3

SIP NOTIFY problem in "presence_dialoginfo" module?
by Klaus Feichtinger 04 Apr '11

04 Apr '11

Hello list, I have a special situation in which string characters of the "root-line" of the notify-body are overwritten by Kamailio. In detail: the root-line of the NOTIFY message sent to the subscriber looks like: <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" ll" entity="sip:117090@10.16.10.99"> instead of <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" state="full" entity="sip:117090@10.16.10.99"> In other words: the string state="full" is overwritten by space characters. Therefore the message is no longer "valid" and the parser of the subscriber-user agent has problems.... The exact scenario in my use case is, that user agents can subscribe the events "presence" and "dialog" on Kamailio. The information about presence and dialog-states is published to Kamailio by PUBLISH messages from the user agents themselves. Kamailio does not have to generate this information itself. In the following lines I have an excerpt of the signalisation: 1) initial subscription of party A, 2) the responded notification (without body) 3) subscription of party B 4) the responded notification (with body) 5) a publish message of the subscribed party B with actual dialog information and 6) the notification with body to the subscriber A. 1) Initial SUBSCRIBE message of party A: ======================================SUBSCRIBE sip:117090@10.16.10.99 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.93:5060;rport;branch=z9hG4bK1411449981 From: <sip:117093@10.16.10.99>;tag=2453899634-24035392-1301468026894 To: <sip:117090@10.16.10.99> Call-ID: 2442490492-24035392-1301468026894(a)10.16.10.93 CSeq: 20 SUBSCRIBE Contact: <sip:117093@10.16.10.93:5060> Max-Forwards: 70 User-Agent: SipTapi Expires: 3600 Event: dialog Content-Length: 0 2) Responded NOTIFY message from Kamailio: ========================================NOTIFY sip:117093@10.16.10.93:5060 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.99;branch=z9hG4bK7cc7.08e060b6000000000000000000000000.0 To: sip:117093@10.16.10.99;tag=2453899634-24035392-1301468026894 From: sip:117090@10.16.10.99;tag=a23c71953194c2c72e41c4d20e4f7127-86e1 CSeq: 1 NOTIFY Call-ID: 2442490492-24035392-1301468026894(a)10.16.10.93 Content-Length: 0 User-Agent: kamailio (3.1.2 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:10.16.10.99:5060> Subscription-State: active;expires=3670 3) Initial SUBSCRIBE message of party B: ======================================SUBSCRIBE sip:117093@10.16.10.99 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.90:5060;rport;branch=z9hG4bK4016956575 From: <sip:117090@10.16.10.99>;tag=2605346227-26329880-1301468174702 To: <sip:117093@10.16.10.99> Call-ID: 2982039389-26329880-1301468174702(a)10.16.10.90 CSeq: 20 SUBSCRIBE Contact: <sip:117090@10.16.10.90:5060> Max-Forwards: 70 User-Agent: SipTapi Expires: 3600 Event: dialog Content-Length: 0 4) Responded NOTIFY message from Kamailio (with version number "00000000000"!!!!): ==============================================================================NOTIFY sip:117090@10.16.10.90:5060 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.99;branch=z9hG4bK10e8.31cb2284000000000000000000000000.0 To: sip:117090@10.16.10.99;tag=2605346227-26329880-1301468174702 From: sip:117093@10.16.10.99;tag=a23c71953194c2c72e41c4d20e4f7127-5f56 CSeq: 1 NOTIFY Call-ID: 2982039389-26329880-1301468174702(a)10.16.10.90 Content-Length: 593 User-Agent: kamailio (3.1.2 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:10.16.10.99:5060> Subscription-State: active;expires=3670 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="00000000000" state="full" entity="117093(a)10.16.10.99"> <dialog id="3375446242-26591296-1301467877329(a)10.16.10.93" call-id="3375446242-26591296-1301467877329(a)10.16.10.93" direction="initiator"> <state>terminated</state> <remote> <identity>sip:019992116002@10.16.10.99</identity> <target uri="sip:019992116002@10.16.10.99"/> </remote> <local> <identity>sip:117093@10.16.10.99</identity> <target uri="sip:117093@10.16.10.99"/> </local> </dialog> </dialog-info> 5) The subscribed party is sending a PUBLISH message with following information: ============================================================================PUBLISH sip:117090@10.16.10.99 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.90:5060;rport;branch=z9hG4bK348084694 From: <sip:117090@10.16.10.99>;tag=3644648667-26329880-1301468223499 To: <sip:117090@10.16.10.99> Call-ID: 332031778-26329880-1301468223499(a)10.16.10.90 CSeq: 20 PUBLISH Max-Forwards: 70 User-Agent: SipTapi Content-Disposition: render;handling=required Expires: 3600 Event: dialog Content-Type: application/dialog-info+xml Content-Length: 592 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="4" state="full" entity="sip:117090@10.16.10.99"> <dialog id="2769397361-26329880-1301468223108(a)10.16.10.90" call-id="2769397361-26329880-1301468223108(a)10.16.10.90" direction="initiator"> <state>confirmed</state> <remote> <identity><sip:116001@10.16.10.99></identity> <target uri="<sip:116001@10.16.10.99>"/> </remote> <local> <identity>sip:117090@10.16.10.99</identity> <target uri="sip:117090@10.16.10.99"/> </local> </dialog> </dialog-info> 6) Status informative NOTIFY message from Kamailio (built from the PUBLISH message above - with correct version but string overwriting!!!): =======================================================================================================================================NOTIFY sip:117093@10.16.10.93:5060 SIP/2.0 Via: SIP/2.0/UDP 10.16.10.99;branch=z9hG4bK4cc7.faa8dc87000000000000000000000000.0 To: sip:117093@10.16.10.99;tag=2453899634-24035392-1301468026894 From: sip:117090@10.16.10.99;tag=a23c71953194c2c72e41c4d20e4f7127-86e1 CSeq: 2 NOTIFY Call-ID: 2442490492-24035392-1301468026894(a)10.16.10.93 Content-Length: 592 User-Agent: kamailio (3.1.2 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:10.16.10.99:5060> Subscription-State: active;expires=3570 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" ll" entity="sip:117090@10.16.10.99"> <dialog id="2769397361-26329880-1301468223108(a)10.16.10.90" call-id="2769397361-26329880-1301468223108(a)10.16.10.90" direction="initiator"> <state>confirmed</state> <remote> <identity><sip:116001@10.16.10.99></identity> <target uri="<sip:116001@10.16.10.99>"/> </remote> <local> <identity>sip:117090@10.16.10.99</identity> <target uri="sip:117090@10.16.10.99"/> </local> </dialog> </dialog-info> Debug information of Kamailio and information in the source code show that the version number "00000000000" is a placeholder only, which should be replaced with a valid version number. However, it seems that there is a problem in handling these strings. The version number is replaced, but in case that the version id is only one character long, ten space characters are inserted- overwriting other information...... Can anybody comment this problem? Is this a bug of the presence_dialoginfo- or another module? Thanks in advance! Regards, Klaus P.S. additional information from source code an debug output of kamailio: notify_body.c: [...] /* The version must be increased for each new document and is a 32bit int. As the version is different for each watcher, we can not set here the correct value. Thus, we just put here a placeholder which will be replaced by the correct value in the aux_body_processing callback. Thus we have CPU intensive XML aggregation only once and can use quick search&replace in the per-watcher aux_body_processing callback. We use 11 chracters as an signed int (although RFC says unsigned int we use signed int as presence module stores "version" in DB as signed int) has max. 10 characters + 1 character for the sign */ [...] Debug information from Kamailio: Ad 4) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:67]: [pres_user]=117093 [pres_domain]= 10.16.10.99, [n]=1 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:107]: [pres_user]=117093 [pres_domain]= 10.16.10.99, [n]=1 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:199]: node type: Element, name: dialog SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:73]: [n_body]=0x835fe54 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:76]: [*n_body]=<?xml version="1.0"?>#012<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="00000000000" state="full" entity="117093(a)10.16.10.99">#012 <dialog id="6b1de4df-6c9ce978(a)10.16.10.203" call-id="6b1de4df-6c9ce978(a)10.16.10.203" direction="recipient">#012 <state>confirmed</state>#012 <remote>#012 <identity>sip:116001@10.16.10.99</identity>#012 <target uri="sip:116001@10.16.10.99"/>#012 </remote>#012 <local>#012 <identity>sip:117093@10.16.10.99</identity>#012 <target uri="sip:117093@10.16.10.99"/>#012 </local>#012 </dialog>#012</dialog-info>#012 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:328]: replace version with "1" SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1461]: dialog info: SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:122]: #012#011[pres_uri]= sip:117093@10.16.10.99#012#011[to_user]117093#011[to_domain]= 10.16.10.99#012#011[w_user]= 117090#011[w_domain]10.16.10.99#012#011[event]= dialog#012#011[status]= active#012#011[expires]3519#012#011[callid]3119346140-8635704-1301477522967(a)10.16.10.90#011[local_cseq]=1#012#011[to_tag ]= a23c71953194c2c72e41c4d20e4f7127-7529#011[from_tag]2282475328-8635704-1301477522967#012#011[contact]sip:117090@10.16.10.90:5060#011[record_route]SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:234]: expires 3519 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1555]: headers:#012Max-Forwards: 70#015#012Event: dialog#015#012Contact: <sip:10.16.10.99:5060>#015#012Subscription-State: active;expires=3570#015#012Content-Type: application/dialog-info+xml#015#012 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:940]: CONTACT sip:117090@10.16.10.90:5060 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 11==11 && [10.16.10.99] =[10.16.10.99] SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1729]: ==22/6/37 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: tm [uac.c:240]: DEBUG:tm:t_uac: next_hop=<sip:117090@10.16.10.90:5060> SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: tm [uac.c:181]: DEBUG: dlg2hash: 27355 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: INFO: presence [notify.c:1586]: NOTIFY sip:117090@10.16.10.99 via sip:117090@10.16.10.90:5060 on behalf of sip:117093@10.16.10.99 for event dialog Ad 6) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:67]: [pres_user]=117093 [pres_domain]= 10.16.10.99, [n]=1 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:107]: [pres_user]=117093 [pres_domain]= 10.16.10.99, [n]=1 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:199]: node type: Element, name: dialog SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:73]: [n_body]=0x835fcb0 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:76]: [*n_body]=<?xml version="1.0"?>#012<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="00000000000" state="full" entity="117093(a)10.16.10.99">#012 <dialog id="6b1de4df-6c9ce978(a)10.16.10.203" call-id="6b1de4df-6c9ce978(a)10.16.10.203" direction="recipient">#012 <state>terminated</state>#012 <remote>#012 <identity>sip:116001@10.16.10.99</identity>#012 <target uri="sip:116001@10.16.10.99"/>#012 </remote>#012 <local>#012 <identity>sip:117093@10.16.10.99</identity>#012 <target uri="sip:117093@10.16.10.99"/>#012 </local>#012 </dialog>#012</dialog-info>#012 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence_dialoginfo [notify_body.c:328]: replace version with "2" SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1461]: dialog info: SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:122]: #012#011[pres_uri]= sip:117093@10.16.10.99#012#011[to_user]117093#011[to_domain]= 10.16.10.99#012#011[w_user]= 117090#011[w_domain]10.16.10.99#012#011[event]= dialog#012#011[status]= active#012#011[expires]3242#012#011[callid]3119346140-8635704-1301477522967(a)10.16.10.90#011[local_cseq]=2#012#011[to_tag ]= a23c71953194c2c72e41c4d20e4f7127-7529#011[from_tag]2282475328-8635704-1301477522967#012#011[contact]sip:117090@10.16.10.90:5060#011[record_route]= SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:234]: expires 3242 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1555]: headers:#012Max-Forwards: 70#015#012Event: dialog#015#012Contact: <sip:10.16.10.99:5060>#015#012Subscription-State: active;expires=3270#015#012Content-Type: application/dialog-info+xml#015#012 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:940]: CONTACT sip:117090@10.16.10.90:5060 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [socket_info.c:501]: grep_sock_info - checking if host==us: 11==11 && [10.16.10.99] =[10.16.10.99] SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: presence [notify.c:1729]: ==22/6/37 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: tm [uac.c:240]: DEBUG:tm:t_uac: next_hop=<sip:117090@10.16.10.90:5060> SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: tm [uac.c:181]: DEBUG: dlg2hash: 27356 SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) SipSrv1 /usr/sbin/kamailio[26764]: INFO: presence [notify.c:1586]: NOTIFY sip:117090@10.16.10.99 via sip:117090@10.16.10.90:5060 on behalf of sip:117093@10.16.10.99 for event dialog

3 6

Suggestion dialog module / $dlg_ctx() problem.
by Alexandre Abreu 04 Apr '11

04 Apr '11

Hello. Using K 3.1.2. Before the setflag() that creates the dialog, I tried to use: $dlg_ctx(flags) = 1; Mar 30 12:29:03 devel kamailio: ERROR: dialog [dlg_var.c:165]: unknown PV name flags Mar 30 12:29:03 devel kamailio: ERROR: <core> [pvapi.c:565]: pvar "dlg_ctx" has an invalid name param [flags] Mar 30 12:29:03 devel kamailio: ERROR: <core> [pvapi.c:720]: wrong char [)/41] in [$dlg_ctx(flags)] at [14 (5)] Mar 30 12:29:03 devel kamailio: : <core> [cfg.y:3409]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 308, column 9-23: unknown script pseudo variable $dlg_ctx(flags) Right now, my goal is to add the user-agent content header ($ua) from caller/callee and list them through 'dlg_list_ctx'. I would suggest something like that: Request route: $dlg_ctx(set->ua_src) = $ua; On reply route: $dlg_ctx(set->ua_dst) = $ua; dlg_list_ctx: . ua_src:: X-Lite 3.0 build 666 ua_dst:: Linksys PAP2 . Does this suggestion make sense? Using another approach is it possible to execute this today? Thanks, Alexandre

4 11

preparing kamailio v3.1.3
by Daniel-Constantin Mierla 04 Apr '11

04 Apr '11

Hello, it is almost 2 months since we packaged 3.1.2, there are commits accumulated in branch 3.1, therefore I plan to package 3.1.3 Monday or Tuesday next week. If anyone has something to add in this regards, please reply. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 1

library dir for x86_64 - lib or lib64
by Daniel-Constantin Mierla 04 Apr '11

04 Apr '11

Hello, iirc this was at some point discussed, however, I want to go through it again. Many x86_64 distros expect libraries to be placed under a lib64 directory. For example the RPM spec file language returns /usr/lib64 for %{_libdir} variable. In the master branch I committed a patch that does this detection. As I saw some people saying that 'lib' should be the default also for x86_64, my question is then what do you think we should use? Using 'lib' will definitely make building rpms fail unless we do a patch specific for these distros. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

3 2

meeting in thailand bangkoke
by Meftah Tayeb 03 Apr '11

03 Apr '11

hi guys i'm in thailand right now this week i'm staying in bangkoke anyone want to meet please drop me an email thank you -- Meftah Tayeb inum: +883510001288000 phone: +13477595883

1 0

Setting Global Variables
by Nathaniel L Keeling 02 Apr '11

02 Apr '11

Hello, I am trying to use global variables within my configuration script. I am using Kamailio v3.0.0 and I am following the syntax example written in the sample config file but I keep getting syntax errors. Here is a copy of my syntax and I am using cfg_get.voicemailserver.bindip to get the value. #!ifdef WITH_VOICEMAIL voicemailserver.bindip = "10.0.0.1" desc "Voicemail Server IP Address" voicemailserver.bindport = "5060" desc "Voicemail Server Port" #!endif Here is the error that I am receiving: 0(1875) : <core> [cfg.y:3333]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 17: syntax error 0(1875) : <core> [cfg.y:3333]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 17: unknown config variable 0(1875) : <core> [cfg.y:3330]: parse error in config file /usr/local/kamailio-3.0.0/etc/kamailio/kamailio.cfg, line 156, column 19-28: ERROR: bad config file (3 errors) 0(1875) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy Thanks Nathaniel

1 0

Avpops problem - bad avp flags.
by Ricardo Martinez 01 Apr '11

01 Apr '11

Hello. I’m having a problems using the avpops with the kamailio version 3.1.2 I have the next configuration: modparam("avpops", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") ----- route[REQINIT] { if( !(avp_db_load("pike", "s:ip_origen") && avp_check("s:ip_origen", "eq/$src_ip/gi")) ) { … } mysql> select * from usr_preferences; +----+------+----------+--------+-----------+------+---------------+---------------------+ | id | uuid | username | domain | attribute | type | value | last_modified | +----+------+----------+--------+-----------+------+---------------+---------------------+ | 1 | pike | 0 | | ip_origen | 0 | 100.20.100.20 | 1900-01-01 00:00:01 | +----+------+----------+--------+-----------+------+---------------+---------------------+ 1 row in set (0.00 sec) When I try to start kamailio I’m getting the next error : Apr 1 10:36:08 kamailio[15418]: ERROR: avpops [avpops_parse.c:129]: error - bad avp flags Apr 1 10:36:08 kamailio[15418]: ERROR: avpops [avpops.c:299]: parse failed Apr 1 10:36:08 kamailio[15418]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:238 Apr 1 10:36:08 kamailio[15418]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:263 I can’t find the problem. Can someone help me? Regards. Ricardo.-

2 1

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