- sr-users - kamailio.org

New developer: Jorj Bauer
by Daniel-Constantin Mierla 11 Aug '14

11 Aug '14

Hello, I want to announce that a new developer got GIT write access to repository: Jorj Bauer from University of Pennsylvania, USA - he is mainly joining the efforts to maintain and develop further the SCA module: * http://kamailio.org/docs/modules/stable/modules/sca.html His git commit id is: jorj My warm welcome and thanks for joining to help the project. Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

Routing for case of client's changing networks
by Dmytro Bogovych 11 Aug '14

11 Aug '14

Greetings all. I have following use case: 0) Peer A and B registers on kamailio via TLS. 1) Peer A establishes call to peer B. 2) Peer A changes network and call has to be refreshed. Peer A reregisters, gathers ICE candidates, builds reINVITE and sends it to B. After peer A changes network again, reregisters and waits for answer from old reINVITE. 3) Peer B receives reINVITE and sends answer. The problem is this answer never gets to peer B. Kamailio cannot route answer as old TLS connection is closed and new one did not established before reregistering. Is there way to solve or avoid this situation? Thank you :)

2 4

http_query results question
by Jonathan Hunter 11 Aug '14

11 Aug '14

Hi Guys, I have used the http_query function in the past, in conjunction with the json module, on version 3.3 and unbuntu. I am now using version 4.1 on Centos 6.5, and I am using the http_query, which I'm aware records in the results the first line of the response. Can you select the result to be recorded depending on the line in the response, for example if a number is returned only in line 4, can this be extracted from a 200 ok response using this function? Or is this then a need to use JSON module, and if so which repository for centos 6.5 contains it! :) Sorry for the question! Thanks Jon

3 8

Dispatcher caches IP
by Yuriy Gorlichenko 11 Aug '14

11 Aug '14

Hello. I try to integrate dispathcer module (with db table dispatcher) to my Kamailio server. I have problem with changing IP at db table, but when I calling - call goes to old IP that alread deleted from db. I tested my table with query that selects destination address from table, and it replyes to me correct address. I use round-robin ("4") algorythm modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:proxy@10.0.0.2") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3") How to disable this cache?

4 5

Accessing Message Buffer in Kamailio
by AliReza Khoshgoftar Monfared 08 Aug '14

08 Aug '14

Hi list, Is there a way to survey the message buffer (iei the queue where arriving sip messages wait to be processed by worker threads) in Kamailio? Is there a way to check the number of messages in this buffer? Additionally, is it possible to survey the number of messages by method (INVITE, BYE, etc.)? I know that the ratelimit module ( http://kamailio.org/docs/modules/stable/modules/ratelimit.html) shall access such buffers, but I do not see any variable accessible in the config offered by that module. Am I missing sth or is there a way to do so? Thanks Alireza

2 2

[Kamailio-4.1.4]Make Install Error
by Djamel Bahamid 08 Aug '14

08 Aug '14

Hello everybody, I am new on this environment ( Kamailio). I try to install(settle) Kamailio 4.1.4, on an O.S Suse Sles11 SP3, all the compilation took place, with the exception of the installation (make install), where I obtain the following error: make[2]: Nothing to be done for `install-if-newer'. touch /usr/local/lib64/kamailio/modules/userblacklist.so install -m 755 userblacklist.so /usr/local/lib64/kamailio/modules make[2]: `libkmi.so.1.0' is up to date. make[2]: `libsrdb1.so.1.0' is up to date. make[2]: `libkcore.so.1.0' is up to date. make[2]: `libsrutils.so.1.0' is up to date. LD (gcc) [M usrloc.so] usrloc.so make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. make[2]: Nothing to be done for `install-if-newer'. touch /usr/local/lib64/kamailio/modules/usrloc.so install -m 755 usrloc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp.so install -m 755 xhttp.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp_rpc.so install -m 755 xhttp_rpc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xlog.so install -m 755 xlog.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xprint.so install -m 755 xprint.so /usr/local/lib64/kamailio/modules /bin/sh: -c: line 18: syntax error near unexpected token `fi' */bin/sh: -c: line 18: ` fi ; \' make: *** [install-cfg] Error 2 * Could you help me please to solve this error, Best Regards, Djamel.

4 14

ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "df7jal23ls0d.invalid"
by Manuel Camarg 08 Aug '14

08 Aug '14

I have this config: WebRTC sipml5 -> Kamailio -> Asterisk -> Kamailio -> WebRTC sipml5 When placing a communication between two sipml5 points I get this errors: ERROR: <core> [resolve.c:1733]: sip_hostport2su(): ERROR: sip_hostport2su: could not resolve hostname: "df7jal23ls0d.invalid" ERROR: <core> [forward.c:532]: forward_request(): ERROR: forward_request: bad host name df7jal23ls0d.invalid, dropping packet ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) If I place a comm between WebRTC sipml5 -> Kamailio -> Asterisk -> Softphone (connected to Asterisk) Everything works fine I'm running RTPEngine as RTP Proxy for DTLS/SRTP issues SIP INVITE signalling Ringing 180 message and 200 OK gets done fine Any ideas where this issue might come from? Looks something related with media traffic (RTP). Kind regards!

2 2

Complex NAT Scenario,HELP!
by Agiftel 08 Aug '14

08 Aug '14

Hi all, I need help configuring a little bit complex NAT/PAT scenario. Idea is that clients on Internet registers themselvs against kamailio and then it routes requests to Alcatel PBX. kamailio acts also as NAT proxy ( so that clients on internet does not need vpn connection. Something like EDGE SERVER for MS Lync clients) Below the scenario Alcatel PBX: 10.9.6.3 Kamailio internal NIC: 172.16.52.240 Kamailio DMZ NIC: 10.9.23.180 PU.BL.I.C.IP: 1.2.3.4 ALCATEL PBX(LAN 10.9.6.3)————-------------- | |———-(LAN INT 172.16.52.240) | |KAMAILIO (2 NICs) ———————————— | | | | |————(LAN dmz 10.9.23.180) PAT/NAT | |————firewal | | PU.BL.I.C.IP ^ | | Client on internet As you can see there is a firewall that make NAT and PAT from PU.BL.I.C.IP to internal Kamailio DMZ nic. natted ports are 5060 and range from 30000 to 65000 ( for RTP ) I'am tryng to use kamailio.cfg that comes with installation.. Now I am making some changes but is not working. Internal client ( lan 172.16.52.x ), registered on kamailio, can make call to alcatel phones and viceversa. External client ( on internet ) can register to kamailio but cannot call alcatel phone nor client registered on kamailio. Here some traces CALL FROM EXTERNAL CLIENT TO INTERNAL CLIENT: *U 2014/07/25 11:26:55.709196 2.2.2.2:39464 -> 10.9.23.180:5060 INVITE sip:1001@lwsbc.longwave.eu SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:casa@2.2.2.2:39464>. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4.6.1 73073-dadc6769-W. Content-Length: 275. . v=0. o=- 13050753989744621 1 IN IP4 192.168.1.102. s=X-Lite release 4.6.1 stamp 73073. c=IN IP4 2.2.2.2. t=0 0. m=audio 65220 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:26:55.822346 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 100 LW-SBC is Trying. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Server: LW-SBC. Content-Length: 0. . U 2014/07/25 11:26:55.891270 172.16.52.240:5060 -> 172.16.52.53:1384 INVITE sip:bria@172.16.52.53:1384;rinstance=b31d64b3d4ead28c SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Max-Forwards: 69. Contact: <sip:casa@2.2.2.2:39464>. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4.6.1 73073-dadc6769-W. Content-Length: 510. . v=0. o=- 13050753989744621 1 IN IP4 192.168.1.102. s=X-Lite release 4.6.1 stamp 73073. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30108 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30109. a=ice-ufrag:wa6qnn48. a=ice-pwd:L2qWms8nR1SA9Z8rWtDiMpzjyEHV. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30108 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30109 typ host. U 2014/07/25 11:26:55.973284 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. To: <sip:1001@lwsbc.longwave.eu>. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Content-Length: 0. . U 2014/07/25 11:26:55.986747 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. User-Agent: Bria 3 release 3.5.5 stamp 71243. Allow-Events: hold, talk. Content-Length: 0. . U 2014/07/25 11:26:56.040076 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. User-Agent: Bria 3 release 3.5.5 stamp 71243. Allow-Events: hold, talk. Content-Length: 0. . U 2014/07/25 11:26:58.473501 10.9.23.180:5060 -> 2.2.2.2:39464 OPTIONS sip:2.2.2.2:39464 SIP/2.0. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0. From: sip:pinger@lwsbc.eu;tag=uloc-53cfd16f-38c-1-b7eb1a5f-72d84677. To: sip:2.2.2.2:39464. Call-ID: b2033011-c4175f65-f074de1(a)10.9.23.180. CSeq: 1 OPTIONS. Content-Length: 0. . U 2014/07/25 11:26:58.551746 2.2.2.2:39464 -> 10.9.23.180:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0;received=1.1.1.1. Contact: <sip:2.2.2.2:39464>. To: <sip:2.2.2.2:39464>;tag=0867c967. From: <sip:pinger@lwsbc.eu>;tag=uloc-53cfd16f-38c-1-b7eb1a5f-72d84677. Call-ID: b2033011-c4175f65-f074de1(a)10.9.23.180. CSeq: 1 OPTIONS. Accept: application/sdp. Accept-Language: en. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: replaces, eventlist. User-Agent: X-Lite release 4.6.1 stamp 73073 52ef741c-W. Content-Length: 0. . U 2014/07/25 11:27:01.110665 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:01.152416 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:01.611529 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:01.649682 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:02.563132 172.16.52.53:1384 -> 172.16.52.240:5060 . . .............. U 2014/07/25 11:27:02.612548 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:02.629843 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:04.613706 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:04.617573 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 535. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 1.1.1.1. t=0 0. a=ice-lite. m=audio 30110 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=rtcp:30111. a=ice-ufrag:ZL2KNsE9. a=ice-pwd:UMuLNnkUmqkiRVyDj6WRKCMJnOr9. a=candidate:Kw2s2DW6WXbD5UKN 1 UDP 2130706432 1.1.1.1 30110 typ host. a=candidate:Kw2s2DW6WXbD5UKN 2 UDP 2130706431 1.1.1.1 30111 typ host. U 2014/07/25 11:27:08.614089 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:08.615487 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:12.614697 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:12.616644 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:14.950640 2.2.2.2:39464 -> 10.9.23.180:5060 . . .............. U 2014/07/25 11:27:16.615213 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:16.617182 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:20.615928 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:20.617554 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:24.616354 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:24.617435 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:28.478391 10.9.23.180:5060 -> 2.2.2.2:39464 OPTIONS sip:2.2.2.2:39464 SIP/2.0. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0. From: sip:pinger@lwsbc.eu;tag=uloc-53cfd16f-38c-1-b7eb1a5f-82d84677. To: sip:2.2.2.2:39464. Call-ID: b2033011-d4175f65-d274de1(a)10.9.23.180. CSeq: 1 OPTIONS. Content-Length: 0. . U 2014/07/25 11:27:28.584706 2.2.2.2:39464 -> 10.9.23.180:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.9.23.180:5060;branch=0;received=1.1.1.1. Contact: <sip:2.2.2.2:39464>. To: <sip:2.2.2.2:39464>;tag=86694f0d. From: <sip:pinger@lwsbc.eu>;tag=uloc-53cfd16f-38c-1-b7eb1a5f-82d84677. Call-ID: b2033011-d4175f65-d274de1(a)10.9.23.180. CSeq: 1 OPTIONS. Accept: application/sdp. Accept-Language: en. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: replaces, eventlist. User-Agent: X-Lite release 4.6.1 stamp 73073 52ef741c-W. Content-Length: 0. . U 2014/07/25 11:27:28.617317 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:28.618441 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:32.564887 172.16.52.53:1384 -> 172.16.52.240:5060 . . .............. U 2014/07/25 11:27:32.618675 172.16.52.53:1384 -> 172.16.52.240:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKa2a3.f650fc43a494d751a41b895338b14212.0;received=172.16.52.240. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2014/07/25 11:27:32.620366 10.9.23.180:5060 -> 2.2.2.2:39464 SIP/2.0 200 OK. Via: SIP/2.0/UDP 2.2.2.2:39464;branch=z9hG4bK-d8754z-015930358feee765-1---d8754z-;rport=39464. Record-Route: <sip:1.1.1.1;lr;r2=on;ftag=a3fb3269;nat=yes>. Record-Route: <sip:10.9.23.180;r2=on;lr=on;ftag=a3fb3269;nat=yes>. Contact: <sip:bria@172.16.52.53:1384>. To: <sip:1001@lwsbc.longwave.eu>;tag=199d4711. From: "CASA"<sip:casa@lwsbc.longwave.eu>;tag=a3fb3269. Call-ID: OTE2ZTRhNDU1NTU4NjU1MzU3MDZlYWUwN2E3MTRjYzk. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces, eventlist. User-Agent: Bria 3 release 3.5.5 stamp 71243. Content-Length: 300. . v=0. o=- 1406280399954939 3 IN IP4 172.16.52.53. s=Bria 3 release 3.5.5 stamp 71243. c=IN IP4 172.16.52.53. t=0 0. m=audio 51960 RTP/AVP 125 100 0 9 8 101. a=rtpmap:125 opus/48000/2. a=fmtp:125 useinbandfec=1. a=rtpmap:100 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. * KAMAILIO CODE: cfg: #!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_DEBUG #!define WITH_ACCDB #!define WITH_XMLRPC #!define WITH_MULTIDOMAIN #!define WITH_ALIASDB #!define WITH_ALCATEL ##------!define WITH_PSTN #!define WITH_XHTTP #!define WITH_ANTIFLOOD #------#!define WITH_WEBSOCKETS #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif include_file "modules_lw.cfg" ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # sip_trace(); # setflag(22); #!ifdef WITH_WEBSOCKETS if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } #!endif # per request initial checks route(REQINIT); # NAT detection #!ifdef WITH_ALCATEL if(($rU=~"^25[0-9][0-9]$")) { route(ALCATEL); } #!endif route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # dispatch destinations to ALCATEL OXE route(ALCATEL); # user location service route(LOCATION); route(RELAY); } #!ifdef WITH_WEBSOCKETS route[SETUP_BY_TRANSPORT] { if ($ru =~ "transport=ws") { xlog("L_INFO", "Request going to WS"); rtpproxy_manage("froc+SP"); t_on_reply("REPLY_FROM_WS"); } else if ($proto =~ "ws") { xlog("L_INFO", "Request coming from WS"); rtpproxy_manage("froc-sp"); t_on_reply("REPLY_TO_WS"); } else { xlog("L_INFO", "This is a classic phone call"); rtpproxy_manage("co"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } #!endif route[RELAY] { #!ifdef WITH_WEBSOCKETS route(SETUP_BY_TRANSPORT); #!endif # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && is_uri_host_local()) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # 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 exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } xlog("L_NOTICE","#### message flags are: $mf [$mF]\n"); if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ALCATEL # do not auth traffic from ALCATEL - trusted! if(route(ALCATEL)) return; #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || is_from_local()) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!is_from_local() && !is_uri_host_local()) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT xlog("L_NOTICE","####$rm from $fu (IP:$si:$sp) Main Route before ---NAT---\n"); force_rport(); if (!nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); xlog("L_NOTICE","####$rm from $fu (IP:$si:$sp) in Route[NAT] fix_nat-register\n"); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT xlog("####NATMANAGE M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if((src_ip==ALCATEL_LAN) || (src_ip==LONGWAVE_LAN)) { rtpproxy_manage("ie"); xlog("L_NOTICE","####$rm $cs from $fu (IP:$si:$sp) in route[RTPPROXY] RTPproxy with IE Flags\n"); } else { rtpproxy_manage("ei"); xlog("L_NOTICE","####$rm $cs from $fu (IP:$si:$sp) in route[RTPPROXY] RTPproxy with EI Flags\n"); } if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(!is_from_local()) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_NOTICE","####new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies #!ifdef WITH_WEBSOCKETS onreply_route[REPLY_TO_WS] { xlog("L_INFO", "####Reply from softphone: $rs"); if (t_check_status("183")) { change_reply_status("180", "Ringing"); remove_body(); exit; } if(!(status=~"[12][0-9][0-9]")) return; rtpproxy_manage("froc+SP"); route(NATMANAGE); } onreply_route[REPLY_FROM_WS] { xlog("L_INFO", "####Reply from webrtc client: $rs"); if(status=~"[12][0-9][0-9]") { rtpproxy_manage("froc-sp"); route(NATMANAGE); } } #!endif onreply_route[MANAGE_REPLY] { xlog("L_NOTICE","####incoming reply Code: $T_reply_code from IP:$si:$sp \n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } # ====================================================== # Populate CDRs Table of Siremis # ====================================================== route[CDRS] { sql_query("cb","call kamailio_cdrs()","rb"); sql_query("cb","call kamailio_rating('default')","rb"); } # -- Charts route[STATS] { # clean very old records $var(tmc) = $var(tmc) + 1; $var(x) = $var(tmc) mod 144; if($var(x) == 0) sql_query("ca", "delete from statistics where time_stamp<$Ts - 864000", "ra"); # insert values for Kamailio internal statistics sql_query("ca", "insert into statistics (time_stamp,shm_used_size," "shm_real_used_size,shm_max_used_size,shm_free_used_size," "ul_users,ul_contacts) values ($Ts,$stat(used_size)," "$stat(real_used_size),$stat(max_used_size),$stat(free_size)," "$stat(location-users),$stat(location-contacts))", "ra"); # init the values for first execution, compute the diff for the rest if($var(tmc)==1) { $var(rcv_req_diff) = $stat(rcv_requests); $var(fwd_req_diff) = $stat(fwd_requests); $var(2xx_trans_diff) = $stat(2xx_transactions); } else { $var(rcv_req_diff) = $stat(rcv_requests) - $sht(stats=>last_rcv_req); $var(fwd_req_diff) = $stat(fwd_requests) - $sht(stats=>last_fwd_req); $var(2xx_trans_diff) = $stat(2xx_transactions) - $sht(stats=>last_2xx_trans); } # update the values for stats stored in cache (htable) $sht(stats=>last_rcv_req) = $stat(rcv_requests); $sht(stats=>last_fwd_req) = $stat(fwd_requests); $sht(stats=>last_2xx_trans) = $stat(2xx_transactions); # insert values for stats computed in config sql_query("ca", "update statistics set tm_active=$stat(inuse_transactions)," "rcv_req_diff=$var(rcv_req_diff),fwd_req_diff=$var(fwd_req_diff)," "2xx_trans_diff=$var(2xx_trans_diff) where time_stamp=$Ts", "ra"); } # ALCATEL GW routing route[ALCATEL] { #!ifdef WITH_ALCATEL # check if ALCATEL GW IP is defined if (strempty($sel(cfg_get.alcatel.gw_ip))) { xlog("SCRIPT: ALCATEL routing enabled but alcatel.gw_ip not defined\n"); return; } # route to ALCATEL dialed numbers starting with '25xx' # - update the condition to match your dialing rules for ALCATEL routing if(!($rU=~"^25[0-9][0-9]$")) return; # only local users allowed to call if(!is_from_local()) { sl_send_reply("403", "Not Allowed!!!!!"); exit; } if (strempty($sel(cfg_get.alcatel.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.alcatel.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.alcatel.gw_ip) + ":" + $sel(cfg_get.alcatel.gw_port); } set_advertised_address("10.9.23.180"); route(RELAY); exit; #!endif return; } modules: ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 mhomed=1 log_facility=LOG_LOCAL1 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT #tcp_connection_lifetime=3605 tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 server_header="Server: LW-SBC" server_signature=yes user_agent_header="User-Agent: LW-SBC" #!ifdef WITH_XHTTP tcp_accept_no_cl=yes #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - 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 #!define FLB_NATSIPPING 7 #!define FLT_OUT 8 #!define ALCATEL_LAN 10.9.6.0/24 #!define LONGWAVE_LAN 172.16.52.0/24 #!substdef "!MY_IP_ADDR_DMZ!10.9.23.180!g" #!substdef "!MY_IP_ADDR_INT!172.16.52.240!g" #!substdef "!MY_DOMAIN!lwsbc.longwave.eu!g" #!substdef "!MY_WS_PORT!10080!g" #!substdef "!MY_WSS_PORT!10443!g" #!substdef "!ALCATEL_IP!10.9.6.3!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR_DMZ:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR_DMZ:MY_WSS_PORT!g" listen=MY_IP_ADDR_DMZ:5060 advertised_address = "1.1.1.1" listen=MY_IP_ADDR_INT:5060 #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif alias = "MY_DOMAIN:5060" alias = "MY_IP_ADDR_DMZ:5060" alias = "MY_IP_ADDR_INT:5060" alias = "MY_DOMAIN:10080" alias = "MY_IP_ADDR_DMZ:10080" /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 port=10080 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "10.9.6.3" desc "PSTN GW Address" pstn.gw_port = "5060" desc "PSTN GW Port" #!endif #!ifdef WITH_ALCATEL # ALCATEL routing # alcatel.gw_ip = "10.9.6.3" desc "Alcatel OXE GW, Longwave" alcatel.gw_port = "5060" desc "Alcatel OXE GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/i386-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "textopsx.so" #!ifdef WITH_WEBSOCKETS loadmodule "websocket.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.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 "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "siptrace.so" loadmodule "sdpops.so" # -- for siremis CDRs -------------- loadmodule "rtimer.so" loadmodule "sqlops.so" # -- Charts Panel #loadmodule "htable.so" #------- GIA' CARICATO CON L'ANTIFLOOD---------- # -- MI Commands Panel loadmodule "mi_datagram.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy-ng.so" #----loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_XHTTP loadmodule "xhttp.so" #!endif #!ifdef WITH_XHTTP_RPC loadmodule "xhttp_rpc.so" #!endif #!ifdef WITH_XHTTP_PI loadmodule "xhttp_pi.so" #!endif # ----------------- 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) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ------- Modifico il valore di default ( trying -- your call is important to us ) modparam("tm", "auto_inv_100_reason", "LW-SBC is Trying") # ----- 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", 1) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- 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) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy-ng", "rtpproxy_sock", "udp:127.0.0.1:12222") #----modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@lwsbc.eu") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "a=>size=8;autoexpire=920;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif #!ifdef WITH_XHTTP_RPC modparam("xhttp_rpc", "xhttp_rpc_root", "http_rpc") #!endif #!ifdef WITH_XHTTP_PI modparam("xhttp_pi", "xhttp_pi_root", "http_pi") modparam("xhttp_pi", "framework", "/etc/kamailio/pi_framework.xml") #!endif # -- for siremis CDRs -------------- modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@localhost/kamailio") # -- For CHARTS modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=STATS") modparam("sqlops","sqlcon", "ca=>mysql://kamailio:kamailiorw@localhost/kamailio") modparam("htable", "htable", "stats=>size=6;") # -- MI Commands Panel modparam("mi_datagram", "socket_name", "udp:127.0.0.1:8033") # ----- siptrace params ----- modparam("siptrace", "db_url", DBURL) modparam("siptrace", "table", "sip_trace") modparam("siptrace", "trace_flag", 22) modparam("siptrace", "traced_user_avp", "$avp(s:user)") modparam("siptrace", "trace_on", 1) #modparam("siptrace", "trace_sl_acks", 1) #modparam("siptrace", "trace_local_ip", "10.9.23.180:5060") /etc/default/ngcp-rtpengine-daemon RUN_MEDIAPROXY=yes LISTEN_TCP=127.0.0.1:25060 LISTEN_UDP=127.0.0.1:22222 LISTEN_NG=127.0.0.1:12222 ADDRESS=10.9.23.180 ADV_ADDRESS=1.1.1.1 # ADDRESS_IPV6=... # ADV_ADDRESS_IPV6=... TIMEOUT=60 SILENT_TIMEOUT=3600 PIDFILE=/var/run/ngcp-rtpengine-daemon.pid FORK=yes # TOS=184 TABLE=0 NO_FALLBACK=yes PORT_MIN=30000 PORT_MAX=50000 # REDIS=127.0.0.1:6379 # REDIS_DB=1 # B2B_URL=http://127.0.0.1:8090/ LOG_LEVEL=7 Regards -- View this message in context: http://sip-router.1086192.n5.nabble.com/Complex-NAT-Scenario-HELP-tp129226.… Sent from the Users mailing list archive at Nabble.com.

2 5

sdpops and rtpproxy/rtpengine
by Alex Balashov 08 Aug '14

08 Aug '14

Hi, Is it safe to use sdpops and the rtpproxy* modules together, given that the latter modify SDP? In other words, can I prune some codecs first, and then call rtpproxy_offer()? Is it safe? Thanks, -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 4

SDPOPS issue or append_hf
by Igor Potjevlesch 07 Aug '14

07 Aug '14

Hello, I have an issue with the module SDPOPS while using "sdp_keep_codecs_by_name". If the calling party sends only one codec description like: Content-Type: application/sdp Content-Length: 202 v=0 o=UserA 2966746938 1790378070 IN IP4 10.141.0.21 s=Session SDP c=IN IP4 10.141.0.21 t=0 0 m=audio 49152 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 The result of the function "sdp_keep_codecs_by_name("PCMA,PCMU,G729a");" is: Content-Type: application/sdp Content-Length: 170 P-Asserted-Identity: "+0123456789" <sip:+0123456789@sip.tld> v=0 o=UserA 2485672881 3000549892 IN IP4 a.b.c.d s=Session SDP c=IN IP4 a.b.c.d t=0 0 m=audio 40330 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=nortpproxy:yes If I open the capture in Wireshark, the PAI is not in the SDP part, and the end of the capture after "a=rtpmap:8 PCMA/8000" is seen as "Data (18 bytes)". I don't understand why the PAI is inserted within the SDP part. Adding the PAI is done after "sdp_keep_codecs_by_name": if (!is_present_hf("P-Asserted-Identity")) { $var(pai) = $(fU{re.subst,/^0/+33/g}); append_hf("P-Asserted-Identity: \"$var(pai)\" <sip:$var(pai)@$fd>\r\n"); } I guess that this cause my INVITE being dropped by 488 Media Not Acceptable Here. Regards, Igor.

2 6

SIP over WebSocket
by Azamat Ailbayev 07 Aug '14

07 Aug '14

Hello, I have a question regarding to SIP over WebSocket implementation. I have changed kamailio.cfg file accordingly with websocket configurations. However, when I am trying to run Kamailio, now I am getting 3 errors on the same line which is "syn_branch = 0". What could be the issue? And How it is possible to fix it? Thank you in advance. BR, Azamat Ailbayev, Kazakhstan

2 1

planning the release of v4.1.5
by Daniel-Constantin Mierla 07 Aug '14

07 Aug '14

Hello, I am considering to release v4.1.5 sometime next week, most likely on the 6th of August. Checking the 4.1 branch, there are not many fixes, few are on my list for backporting. That's good, indicating a high level of stability. If anyone is aware of issues not reported on tracker or patches that have not been backported, add to the tracker or write a message to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 5

Kamailio v4.1.5 Released
by Daniel-Constantin Mierla 06 Aug '14

06 Aug '14

Hello, Kamailio SIP Server v4.1.5 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.4. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.5. For more details about version 4.1.5 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/08/kamailio-v4-1-5-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio with open ims
by Senthil K 06 Aug '14

06 Aug '14

Hi All, i have a query here, the user is registered with the open IMS and also with Kamailio ( as a third party registration via SCSCF ) Here my issue is ; when the user makes a call, the INVITE reaches to kamailio from SCSCF( Scscf port is 6060) , when the Kamailio sends the INVITE back to IMS, it sends to the port 5060. problem is , the kamailio registers the SCSCF port as 6060, but when sending the INVITE back it uses the port 5060. why this is So ?? i checked in registered entry as well, it shows that the contact is SCSCF with 6060 port only. please help in this situation, Kind regards SenthilK

2 2

Kamailio with rtpproxy-ng and mediaproxy-ng: Error rewriting SDP
by Olli Heiskanen 06 Aug '14

06 Aug '14

Hello, I've been experimenting with Kamailio with ws and sip clients and could need a hand in getting a call between those two to work. I have Kamailio 4.1.2 (using rtpproxy-ng instead of rtpproxy) on a CentOS 6.5 and a mediaproxy-ng running. I have clients wsclient(a)testers.com and gsclient(a)testers.com and I try to make call from wsclient to gsclient. The wsclient is a jssip client running on chrome and gsclient is a grandstream desk phone. My config file is the default one enhanced by online examples. I use a html5 <audio> element for the media streams, and configured my jssip phone to accept audio options like this: var options = { 'eventHandlers': eventHandlers, 'mediaConstraints': {'audio': true, 'video': false } }; sipUA.call(callto, options); I used the instructions from here: http://www.slideshare.net/crocodilertc/webrtc-websockets What I get is gsclient ringing, and as I answer there is no audio and call hangs up in a few seconds. I guess this is a SDP problem, something between Kamailio and Mediaproxy-ng but SDP is not my strong point so I'd appreciate advice. Question is where's my misconfiguration/problem? I would like to learn why this problem occurs and how to fix it rather than getting a solution right away, but please bear in mind I don't know much about SDP. In Kamailio log I see: kamailio[27059]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Error rewriting SDP kamailio[27058]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Unknown call-id kamailio[27057]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Unknown call-id Following are the INVITEs and 200 OKs from my SIP trace (1.1.1.1 is the ip of my Kamailio & mediaproxy-ng box and 2.2.2.2 is the public ip behind which both my clients are). The gsclient has port 5066. ****************************************************************************** U 2014/04/01 20:03:41.060009 1.1.1.1:5060 -> 2.2.2.2:5066 INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Max-Forwards: 16. To: <sip:gsclient@testers.com>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. Contact: <sip:wsclient@testers.com ;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>. Allow: ACK,CANCEL,BYE,OPTIONS,INVITE. Content-Type: application/sdp. Supported: path, outbound, gruu. User-Agent: JsSIP 0.3.0. Content-Length: 2211. . v=0. o=- 4897716268503406223 2 IN IP4 1.1.1.1. s=-. t=0 0. a=group:BUNDLE audio. a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ. m=audio 30028 RTP/SAVPF 111 103 104 0 8 106 105 13 126. c=IN IP4 1.1.1.1. a=candidate:2999745851 1 udp 2113937151 192.168.56.1 63341 typ host generation 0. a=candidate:2999745851 2 udp 2113937151 192.168.56.1 63341 typ host generation 0. a=candidate:3350409123 1 udp 2113937151 192.168.0.101 63342 typ host generation 0. a=candidate:3350409123 2 udp 2113937151 192.168.0.101 63342 typ host generation 0. a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0. a=candidate:4233069003 2 tcp 150995 T 2014/04/01 20:03:41.119806 2.2.2.2:38986 -> 1.1.1.1:5060 [A] ...... U 2014/04/01 20:03:41.159086 2.2.2.2:5066 -> 1.1.1.1:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=7875f08763872c34. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Warning: 304 GS "Media type not available". Content-Length: 0. . U 2014/04/01 20:03:41.159392 1.1.1.1:5060 -> 2.2.2.2:5066 ACK sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Max-Forwards: 16. To: <sip:gsclient@testers.com>;tag=7875f08763872c34. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 ACK. Content-Length: 0. . U 2014/04/01 20:03:41.161085 1.1.1.1:5060 -> 2.2.2.2:5066 INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Max-Forwards: 16. To: <sip:gsclient@testers.com>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. Contact: <sip:wsclient@testers.com ;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>. Allow: ACK,CANCEL,BYE,OPTIONS,INVITE. Content-Type: application/sdp. Supported: path, outbound, gruu. User-Agent: JsSIP 0.3.0. Content-Length: 3136. . v=0. o=- 4897716268503406223 2 IN IP4 1.1.1.1. s=-. t=0 0. a=group:BUNDLE audio. a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ. m=audio 30028 RTP/AVP 111 103 104 0 8 106 105 13 126. c=IN IP4 1.1.1.1. a=fingerprint:sha-256 72:54:87:EC:D2:4C:D1:70:C2:FE:69:08:20:5C:92:1D:E0:EA:BD:45:09:E0:90:62:27:B6:34:60:54:E2:99:28. a=setup:actpass. a=mid:audio. a=sendrecv. a=rtpmap:111 opus/48000/2. a=fmtp:111 minptime=10. a=rtpmap:103 ISAC/16000. a=rtpmap:104 ISAC/32000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:106 CN/32000. a=rtpmap:105 CN/16000. a=rtpmap:13 CN/8000. a=rtpmap:126 telephone-event/8000. a=maxptime:60. a=ssrc:3298511848 cnam And here are the 200 OK messages when answering the call: U 2014/04/01 20:03:46.049711 2.2.2.2:5066 -> 1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=fb215901a251c9a0. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Contact: <sip:gsclient@192.168.0.106:5066;transport=udp>. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE. Content-Type: application/sdp. Supported: replaces, timer. Content-Length: 216. . v=0. o=gsclient 8000 8000 IN IP4 192.168.0.106. s=SIP Call. c=IN IP4 192.168.0.106. t=0 0. m=audio 5026 RTP/AVP 0 13. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126. T 2014/04/01 20:03:46.051127 1.1.1.1:5060 -> 2.2.2.2:38986 [AP] .~.dSIP/2.0 200 OK. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>. Record-Route: <sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes>. From: <sip:wsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com>;tag=fb215901a251c9a0. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Contact: <sip:gsclient@192.168.0.106:5066;transport=udp>. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE. Content-Type: application/sdp. Supported: replaces, timer. Content-Length: 216. . v=0. o=gsclient 8000 8000 IN IP4 192.168.0.106. s=SIP Call. c=IN IP4 192.168.0.106. t=0 0. m=audio 5026 RTP/AVP 0 13. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126. ****************************************************************************** cheers, Olli

4 26

no fork mode and more than one listen address found (will use only the first one)
by Yuriy Gorlichenko 06 Aug '14

06 Aug '14

Hello. I Installed kamailio 4.1.4, when I starting my server I see following Warning no fork mode and more than one listen address found (will use only the first one) So I see only first listening port at my netstat At my kamailio.cfg file I write fork=yes? but it not helps me. I found the same issue at this mailing list http://comments.gmane.org/gmane.comp.voip.openser.user/10686 But it very old issue (2007 year) and I think patch at this issue does not help me. How I can start my Server for runnings with many interfaces?

3 5

Understanding DNS resolution done by Kamailio
by Igor Potjevlesch 05 Aug '14

05 Aug '14

Hello, I'm using Kamailio 4.1 and I'm wondering how to avoid external DNS resolution. I have the following config: dns_cache_init=no use_dns_cache=no dns=no rev_dns=no Even with this config, I have many and many DNS query on SRV _sip for the hostnames set in carrierroute module. These requests are not useful because the A resolution is done by /etc/hosts. Major problem with this, is that when I have a DNS issue or IP transit issue, Kamailio waits for resolution timeout and becomes overloaded. As a consequence, Kamailio can't treat others SIP requests like REGISTER because he stuck in DNS resolution. Regards, Igor.

2 4

[rtpengine] continuous 'Successful STUN binding request from' messages
by Paul Belanger 05 Aug '14

05 Aug '14

I was hoping somebody could confirm the following is a _normal_ log file for rtpengine. Specifically I am curious of the 'Successful STUN binding request from' messages are continuously logged. Are they part of the normal operation or does this indicate some issue with media? I only ask because I am getting one way audio at the moment. I get audio from rtpengine however rtpengine does not hear me. http://pastebin.com/hztxMFL3 -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 3

rtp relaying and cseq - rtpengine/rtpproxy
by Daniel-Constantin Mierla 05 Aug '14

05 Aug '14

Hello, apparently none of the rtp relays I was looking at (rtpproxy and rtpenging) are taking in consideration the CSeq value. Quite recently I met a case with a b2bua that sends the invite without sdp, then after its ACK with SDP sends quickly a re-invite to get itself out of media stream. However, due to network and parallel processing races, the ACK gets processed after the re-INVITE, updating the rtp relay session with attributes that should be no longer used. The impact is practically no audio. I got it working using the 'brave' htable and tracking cseq numbers, skipping calling the relay for lower cseqs, but I wonder if wouldn't be better to push it to the relay, to avoid also races on the network between kamailio and the relay. I would be also curious to hear if others were facing such issues. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA

1 0

Modify Refer-To header
by Stefan Kohlhauser 05 Aug '14

05 Aug '14

2 1

Re: [SR-Users] missing BYE record on acc table when usingtimeout_avp
by betul.cerit＠netgsm.com.tr 05 Aug '14

05 Aug '14

Hi, In ‘event_route[dialog:end]’I can not get any variable they are all empty but using acc_db_requests() worked for me. Thanks for your help. BEST REGARDS From: Daniel-Constantin Mierla Sent: Monday, August 04, 2014 5:28 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] missing BYE record on acc table when usingtimeout_avp Hello, with latest version you can get the cdrs at once via dialog+acc module. If you want to use the classic acc, then add either event_route[dialog:end] or event_route[tm:local-request]. There you can get actions when the dialog is ended (including the time out), and for the second you get the BYE sent out. Perhaps SQLOPS with sql_query() or acc_db_requests() should be used to record the event in db table. Cheers, Daniel On 04/08/14 15:19, betul.cerit(a)netgsm.com.tr wrote: Hi to all, I want to limit call duration according to the credit of the user.To accomplish this I use timeout_avp .When the limit is reached call ends but no record associated with BYE method is inserted to the acc table and that makes impossible to bill that call.What can I do to solve this problem or is there an alternative way to limit call duration?My configuration and logs are as follows. BEST REGARDS. #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #endif -------------------------------------------------------------------------------- route[SW]{ xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); $dlg_ctx(timeout_bye)=1; $avp(i:10)=15; $dlg_ctx(timeout_route)=33; dlg_manage(); set_dlg_profile("userid","$avp(userid)"); if(get_profile_size("userid","$avp(s_userid)")){ xlog("L_INFO","Userid_Size:$avp(s_userid)"); } xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); if(!t_relay()) { send_reply("408", "Servis Disi"); }else{ exit; } return; } event_route[dialog:start]{ xlog("L_ALERT","START:ci:$ci:method:$rm:start"); } event_route[dialog:end]{ xlog("L_ALERT","END"); xlog("L_ALERT","END:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_ALERT","END:$rm:$rr:$rs:"); } event_route[dialog:failed]{ xlog("L_INFO","FAILED:ci:$ci:method:$rm:end"); } route[33]{ xlog("L_INFO","route:33:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_INFO","route:33:$rm:$rr:$rs:"); } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } -------------------------------------------------------------------------------- Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: USERID:456123 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: arayan:XXXX:aranan:XXXXX Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri XXXX was rewritten to sip:YYYYYYYYYYYYYYY@XXXXXXX5:5060, carrier 3, domain 1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: vardesc: 40 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:<null>:h_entry:<null>:timeout:0:bye:0:<null> Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Userid_Size:1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:3953:h_entry:3170:timeout:14:bye:1:15 Aug 1 16:51:55 kamailio /usr/local/sbin/kamailio[6114]: ALERT: <script>: START:ci:2c5695c1644fa2b135f57ea72c590cc7@XXXXXXX:5060:method:INVITE:start Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: WARNING: dialog [dlg_req_within.c:212]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7feb2115d918 [3170:3953] with clid '2c5695c1644fa2b135f57ea72c590cc7@95.0.154.92:5060' and tags 'as3c7c5aa4' 'as0b640624' Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: ERROR: acc [acc_cdr.c:574]: cdr_on_end(): invalid values#012! _______________________________________________ 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda -------------------------------------------------------------------------------- _______________________________________________ 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

2 1

[rtpengine] DTLS error: 1 (message digest is null)
by Paul Belanger 04 Aug '14

04 Aug '14

Greetings, I'm having some trouble getting dtls-srtp working with kamailio 4.1 (mediaproxy-ng) and rtpengine (master). I believe I finally have my branch logic setup properly in kamailio, however when the calls get placed into rtpengine, it appears DTLS is not initialized properly. Attached is my debug log file. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 3

4.1 Compilation issue
by Gilbert T. Gutierrez, Jr. 04 Aug '14

04 Aug '14

CentOS6.5 Kamailio 4.1 I compiled Kamailio as follows so that the configuration file would be in "/etc/kamailio" rather than "/usr/local/etc/kamailio/"... make cfg_prefix=/ all make cfg_prefix=/ install When I try to start kamailio I get the following error... Not starting : invalid configuration file! ERROR: loading config file(/usr/local/etc/kamailio/kamailio.cfg): No such file or directory I can find no reference in the init file to that location (which does not exist). Did I use the wrong options in make? Gilbert

3 3

missing BYE record on acc table when using timeout_avp
by betul.cerit＠netgsm.com.tr 04 Aug '14

04 Aug '14

Hi to all, I want to limit call duration according to the credit of the user.To accomplish this I use timeout_avp .When the limit is reached call ends but no record associated with BYE method is inserted to the acc table and that makes impossible to bill that call.What can I do to solve this problem or is there an alternative way to limit call duration?My configuration and logs are as follows. BEST REGARDS. #---------------- dialog params ------------- #!ifdef WITH_DIALOG modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 8192) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag",4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "default_timeout", 3600) modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "userid ; opkodu") modparam("dialog", "dlg_extra_hdrs", "Hint: Hell Yeah\r\n") modparam("dialog", "send_bye", 1) #endif -------------------------------------------------------------------------------- route[SW]{ xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); $dlg_ctx(timeout_bye)=1; $avp(i:10)=15; $dlg_ctx(timeout_route)=33; dlg_manage(); set_dlg_profile("userid","$avp(userid)"); if(get_profile_size("userid","$avp(s_userid)")){ xlog("L_INFO","Userid_Size:$avp(s_userid)"); } xlog("L_INFO","Dialog:h_id:$dlg(h_id):h_entry:$dlg(h_entry):timeout:$dlg_ctx(timeout_route):bye:$dlg_ctx(timeout_bye):$avp(i:10)"); if(!t_relay()) { send_reply("408", "Servis Disi"); }else{ exit; } return; } event_route[dialog:start]{ xlog("L_ALERT","START:ci:$ci:method:$rm:start"); } event_route[dialog:end]{ xlog("L_ALERT","END"); xlog("L_ALERT","END:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_ALERT","END:$rm:$rr:$rs:"); } event_route[dialog:failed]{ xlog("L_INFO","FAILED:ci:$ci:method:$rm:end"); } route[33]{ xlog("L_INFO","route:33:$fU:$fd:$si:$rU:$rd:$avp(poparty):$avp(networkdatetime):$avp(ptparty):$avp(pprovider):$avp(maliyet_id):$avp(tibtip)"); xlog("L_INFO","route:33:$rm:$rr:$rs:"); } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } -------------------------------------------------------------------------------- Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: USERID:456123 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: ALERT: <script>: arayan:XXXX:aranan:XXXXX Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: carrierroute [cr_func.c:710]: cr_do_route(): uri XXXX was rewritten to sip:YYYYYYYYYYYYYYY@XXXXXXX5:5060, carrier 3, domain 1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: vardesc: 40 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:<null>:h_entry:<null>:timeout:0:bye:0:<null> Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Userid_Size:1 Aug 1 16:51:52 kamailio /usr/local/sbin/kamailio[6111]: INFO: <script>: Dialog:h_id:3953:h_entry:3170:timeout:14:bye:1:15 Aug 1 16:51:55 kamailio /usr/local/sbin/kamailio[6114]: ALERT: <script>: START:ci:2c5695c1644fa2b135f57ea72c590cc7@XXXXXXX:5060:method:INVITE:start Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: INFO: <script>: route:33:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse From uri Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ERROR: pv [pv_core.c:397]: pv_get_xto_attr(): cannot parse From URI Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:<null>:<null>:1.0.0.127:you:kamailio.org:<null>:<null>:<null>:<null>:<null>:<null> Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6145]: ALERT: <script>: END:OPTIONS:<null>:<null>: Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: WARNING: dialog [dlg_req_within.c:212]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7feb2115d918 [3170:3953] with clid '2c5695c1644fa2b135f57ea72c590cc7@95.0.154.92:5060' and tags 'as3c7c5aa4' 'as0b640624' Aug 1 16:52:11 kamailio /usr/local/sbin/kamailio[6121]: ERROR: acc [acc_cdr.c:574]: cdr_on_end(): invalid values#012!

2 2

LM_ERR and LM_DBG
by aawaise 04 Aug '14

04 Aug '14

I want to ask some queries regarding logging in kamailio. 1. First of all if I want to add LM_DBG or LM_ERR in lookup.c file of registrar module, where to add the command as on extraction of kamailio-3.1.1.zip, I found registrar module files in modules_k folder as well as modules_s folder. 2. I was advised to us macros from dprint.h to see LM_DBG and LM_ERR statements in log file, how to do so ?? 3. Third, will these logs also print in /var/log/messages in debian linux along with other kamailio.cfg logs ?? 4. What changes should be made in kamailio.cfg to see these logs, i.e log level etc ?? Any help will be highly appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/LM-ERR-and-LM-DBG-tp129392.html Sent from the Users mailing list archive at Nabble.com.

2 1

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