- sr-users - kamailio.org

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ 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

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

usrloc module expires column issues
by Grant Bagdasarian 10 Nov '16

10 Nov '16

Hello, When saving in the location table, the expires and last_modified columns contain the value 000-00-00 00:00:00 which causes issues when Kamailio is restarted and the preload modparam is set to load the location table from db in memory at startup. I've already upgraded Kamailio to 4.4.3, and used the dbtables script from version 4.4 to recreate the location table, but without success. Anyone else having this issue? Any resolution for it? Regards, Grant Bagdasarian CM

2 9

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, Here more clear trace https://www.dropbox.com/s/2z6ck71ulidqelh/kamailio-BYE-flow.png?dl=0 BYE send by kamailio to wrong freeswitch box. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 20:09:53 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, Please check this sip call flow picture https://www.dropbox.com/s/itdewdg3ph7xcyx/kamailio-fs-BYE-flow.gif?dl=0 Kamilio send BYE to incorrect freeswitch which already responded to BYE from leg one. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 19:11:56 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, Please check this sip call flow picture https://www.dropbox.com/s/itdewdg3ph7xcyx/kamailio-fs-BYE-flow.gif?dl=0 Kamilio send BYE to incorrect freeswitch which already responded to BYE from leg one. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 19:11:56 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Kamailio v4.4.4 Released
by Daniel-Constantin Mierla 09 Nov '16

09 Nov '16

Hello, Kamailio SIP Server v4.4.4 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.4. For more details about version 4.4.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/11/kamailio-v4-4-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Count number of early(Ringing) dialogs on gateway
by Aqs Younas 09 Nov '16

09 Nov '16

Greetings list, I am using dialog aur dispatcher modules for calls distribution across multiple gateways. I am able to get number of active dialogs against a gateway but could not get early dialogs going on particular gateway. Dialog information is inserted in database only for state 4. Also, I tried to use dlg_list command to get early dialogs. I see call statistics but "callee_contact::" is empty untill call is answered. for ringing I am getting this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo dlg_list dialog:: hash=3443:9422 state:: 2 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: GGjl58SQ32LXQB1dV2-CBA.. from_uri:: sip:1015@192.168.10.39:5060;transport=UDP from_tag:: 652ee907 caller_contact:: sip:1015@192.168.10.41:55842;transport=UDP caller_cseq:: 1 caller_route_set:: caller_bind_addr:: udp:192.168.10.39:5060 callee_bind_addr:: to_uri:: sip:1010@192.168.10.39:5060;transport=UDP to_tag:: callee_contact:: callee_cseq:: callee_route_set:: callee_contact:: is empty. Is there any way to get early dialog info againts a gateway.? Or any way to insert early dialog into database with callee_contact not empty? Best Regards, Aqs Younas

2 2

Planning the release of Kamailio v4.4.4
by Daniel-Constantin Mierla 09 Nov '16

09 Nov '16

Hello, I am considering to package a new release out of latest stable branch 4.4, respectively v4.4.4. The target date is one of the days between Nov 8-10, a matter of work load that needs to be done for getting it out. As usual, if you are aware of any issue not yet reported on github bug tracker, do it as soon as possible to give it a chance to be fixed: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

1 2

Re: [SR-Users] fnmatch
by Slava Bendersky 09 Nov '16

09 Nov '16

Hello Daniel, Yes, reply was 202 not 200. Right now it work as expected. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, 7 November, 2016 04:51:11 Subject: Re: [SR-Users] fnmatch Hello, for clarification -- what's the actual problem here? fnmatch("$rs", "200") is matching the 202? Cheers, Daniel On 07/11/16 08:52, Serge S. Yuriev wrote: Hello, In debug you can see 202 Accepted but match configured on 200. Not matched -- Wbr, Serge via mobile 07.11.2016, 01:08, "Slava Bendersky" <volga629(a)skillsearch.ca> : BQ_BEGIN Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 3

BYE dispatcher
by Slava Bendersky 09 Nov '16

09 Nov '16

Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). client_public_ip.28972 > proxy_public_ip.llrp: [udp sum ok] UDP, length 660 E....2..5...c....E.\q,.....CBYE sip:mod_sofia@10.18.130.24:5160 SIP/2.0 Via: SIP/2.0/UDP client_public_ip:28972;branch=z9hG4bK1248337638;rport Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=01K2Q1BFa9pKH> Route: <sip:10.18.130.27;r2=on;lr=on;ftag=01K2Q1BFa9pKH> From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE Contact: <sip:4300@client_public_ip:28972> Max-Forwards: 70 Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 IP (tos 0x10, ttl 64, id 10813, offset 0, flags [none], proto UDP (17), length 930) 10.18.130.27.sip > 10.18.130.24.5160: [bad udp cksum 0x1bf7 -> 0xd85c!] UDP, length 902 E...*=..@.4. ... ......(....BYE sip:mod_sofia@10.18.130.24:5160 SIP/2.0 Record-Route: <sip:10.18.130.27;r2=on;lr=on;ftag=487304085> Record-Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=487304085> Via: SIP/2.0/UDP 10.18.130.27;branch=z9hG4bKa61.22917bc5fdd65f16e39bf09dcbc7316b.0 Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=01K2Q1BFa9pKH> Route: <sip:10.18.130.27;r2=on;lr=on;ftag=01K2Q1BFa9pKH> From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE Contact: <sip:4300@client_public_ip:28972> Max-Forwards: 69 Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 IP (tos 0x0, ttl 64, id 38202, offset 0, flags [none], proto UDP (17), length 587) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 559 E..K.:..@... ... ....(...7i.SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.27;branch=z9hG4bKa61.22917bc5fdd65f16e39bf09dcbc7316b.0 Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 IP (tos 0x10, ttl 64, id 782, offset 0, flags [none], proto UDP (17), length 503) proxy_public_ip.llrp > client_public_ip.28972: [bad udp cksum 0x4c50 -> 0xd018!] UDP, length 475 E.......@.+}.E.\c.....q,..LPSIP/2.0 200 OK Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 IP (tos 0x0, ttl 64, id 10584, offset 0, flags [none], proto UDP (17), length 764) 10.18.130.26.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 736 E...)X..@.6@ ... ....(....Y^BYE sip:4200@client_public_ip:48147;rinstance=c4084e18f3f39974;alias=client_public_ip~48147~1 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.26:5160;rport;branch=z9hG4bK33785247SrX3H Route: <sip:10.18.130.27;r2=on;lr=on;ftag=d2e84f4a> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=d2e84f4a> Max-Forwards: 70 From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 802, offset 0, flags [none], proto UDP (17), length 741) proxy_public_ip.llrp > client_public_ip.48147: [bad udp cksum 0x4d3e -> 0x4f2d!] UDP, length 713 E...."..@.*{.E.\c.........M>BYE sip:4200@client_public_ip:48147;rinstance=c4084e18f3f39974 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK8395.d46858df1d9e02b910446fca09f1f864.0 Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H Max-Forwards: 69 From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 810, offset 0, flags [DF], proto UDP (17), length 504) client_public_ip.48147 > proxy_public_ip.llrp: [udp sum ok] UDP, length 476 E....*@.4..oc....E.\........SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK8395.d46858df1d9e02b910446fca09f1f864.0 Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE Accept-Language: en Content-Length: 0 IP (tos 0x10, ttl 64, id 63040, offset 0, flags [none], proto UDP (17), length 413) 10.18.130.27.sip > 10.18.130.26.5160: [bad udp cksum 0x19f4 -> 0x6386!] UDP, length 385 E....@..@.j. ... ......(....SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE Accept-Language: en Content-Length: 0 Slava.

2 1

Count number of early(Ringing) dialogs on gateway
by Aqs Younas 09 Nov '16

09 Nov '16

Greetings list, I am using dialog aur dispatcher modules for calls distribution across multiple gateways. I am able to get number of active dialogs against a gateway but could not get early dialogs going on particular gateway. Dialog information is inserted in database only for state 4. Also, I tried to use dlg_list command to get early dialogs. I see call statistics but "callee_contact::" is empty untill call is answered. For ringing I am getting this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo dlg_list dialog:: hash=3443:9422 state:: 2 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: GGjl58SQ32LXQB1dV2-CBA.. from_uri:: sip:1015@192.168.10.39:5060;transport=UDP from_tag:: 652ee907 caller_contact:: sip:1015@192.168.10.41:55842;transport=UDP caller_cseq:: 1 caller_route_set:: caller_bind_addr:: udp:192.168.10.39:5060 callee_bind_addr:: to_uri:: sip:1010@192.168.10.39:5060;transport=UDP to_tag:: callee_contact:: callee_cseq:: callee_route_set:: callee_contact:: is empty. Is there any way to get early dialog info against each gateway or every dialog stat insertion in database dialog table. Best Regards.

1 0

using set_body_multipart and append_body_part on on_reply routes
by david 08 Nov '16

08 Nov '16

hello all i'm using kam version 4.4.1 and i would like to generate an ISUP part on the 1xx,2xx responses to a remote carrier since the fsw i send the calls to cannot handle ISUP SDP. despite seeing in the doc that those functions cannot be used there and only in REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE, i tried and it works.... partially, because it seems it doesnt create the last boundary and it creates a next boundary with an empty "Encapsulated multipart part: " this the the route i call from onreply_route route[SIPISUP] { set_body_multipart(); msg_apply_changes(); $var(isup) = "\x01\x12\x49\x01\x0a\x03\x02\x0a\x08\x84\x90\x33\x41 \x72\x17\x01\x06\x0a\x08\x04\x13\x93\x70\x21\x73\x23\x10\x08\x01\x01 \x00"; append_body_part("$var(isup)","application/isup;version=itu-t92 +","signal;handling=optional"); msg_apply_changes(); sipt_destination($rU, 31, 4); sipt_set_calling($fU, 4, 0, 3); msg_apply_changes(); } am i doing something wrong? how can i set this to end the SDP with last boundary? thanks a lot and regards david

2 3

Re: [SR-Users] kamailio + farm of Elastix + rtp direct
by Kato 08 Nov '16

08 Nov '16

Any comments on this please? Thanks! De: Kato [mailto:elkato@gmail.com] Enviado el: domingo, 6 de noviembre de 2016 11:29 a. m. Para: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Asunto: kamailio + farm of Elastix + rtp direct Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

1 0

Next Kamailio IRC Devel Meeting, Nov 09, 2016
by Daniel-Constantin Mierla 08 Nov '16

08 Nov '16

Hello, I am proposing a new IRC meeting to discuss the current major issues and the plans for next Kamailio release, on Wednesday, Nov 09, 2016, with alternatives for previous or next day. If many developers are not available, we can postpone it to another date in the near future (make proposals if that is the case for you). I created a wiki page for it: * https://www.kamailio.org/wiki/devel/irc-meetings/2016b Add there the topics that you want to be discussed. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 4

Contacts replacing each other when registering
by Sebastian Damm 08 Nov '16

08 Nov '16

Hi, we are running a Registrar server with 4.4.3, the registrar module is configured with the following parameters: modparam("registrar", "max_contacts", 50) modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("registrar", "received_avp", "$(avp(i:801))") modparam("registrar", "default_expires", 600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 600) modparam("registrar", "gruu_enabled", 0) A customer has configured 4 phones with the same username. Normally, all 4 phones would be visible in the location database, and incoming calls would fork out to all 4 phones. However, only one of the devices is always online, the others always replace each other on each registration. Client details: * All phones are snom 715 devices with the same firmware version. * All phones sit behind the same router. Registrations: client 1: * Via: SIP/2.0/UDP 192.168.208.91:35812;branch=z9hG4bK-4q0mmxgu6gm7;rport * Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 2: * Via: SIP/2.0/UDP 192.168.208.90:41331;branch=z9hG4bK-3cbxdvbmk0di;rport * Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 3: * Via: SIP/2.0/UDP 192.168.208.92:5360;branch=z9hG4bK-4drsogakez94;rport * Contact: <sip:1794388e3@192.168.208.92:5360;line=0n7jfzih>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 4: * Via: SIP/2.0/UDP 192.168.208.93:5260;branch=z9hG4bK-1ci0qia3yxyl;rport * Contact: <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;reg-id=1;q=1.0 Only client 4 is reachable all the time. The other 3 clients replace other registrations. So after client 1 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;q=1;expires=600;received="sip:1.2.3.4:6411";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=303;received="sip:1.2.3.4:8923" And after client 2 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;q=1;expires=600;received="sip:1.2.3.4:49061";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=301;received="sip:1.2.3.4:8923" So you see, one contact stays the same, the other one gets replaced. Does anybody know what could cause this kind of behavior? I think, it could be GRUU. I see that those three clients all send the same GRUU parameter. From my understanding, this UUID should be different from client to client. I don't know whether this is a bug in the snom firmware always sending the same GRUU UUID or whether som "intelligent" SIP aware router is putting this in the Contact header. However, since we have disabled gruu for the registrar module (see above), I think, Kamailio should ignore it. Ideas? Thanks in advance. Best Regards, Sebastian

2 4

TCP Keepalive for NAT'd Clients
by Colin Morelli 08 Nov '16

08 Nov '16

Hey all, Looking to figure out the best way to allow TCP connections to stay alive for NAT'd clients, however, to protect against people just opening TCP connections to the server, I'm hoping to only keepalive TCP connections for connections that have sent an INVITE and received a 200 OK. Does this type of per-socket connection lifetime setting exist? For a bit more background: this is specifically for clients that are using the TCP connection for an INVITE. They don't register with the server, and as a result there's no active flow timer for their connection. Is there a way to enable this type of behavior without registering? Best, Colin

2 2

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 08 Nov '16

08 Nov '16

Hello I am using Kamailio to collect data about active calls. I write INVITEs to database, and delete it on BYE or any REL. I would like to write source and destination port too, but don't know how to get destination port. I have tried variables: $dp, $rp, $Rp, $op But when I don't have port in URI like INVITE IP:PORT, $rp have default value of 5060. I would like to get real UDP port, that INVITE does to. Is it possible using PSEUDO VARIABLES? Greetings Andrzej

2 1

Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB
by Jurijs Ivolga 08 Nov '16

08 Nov '16

Hi, Our Kamailio server is crashing once per week, with following error: Oct 1 06:25:06 kamailio kernel: [26982632.803789] Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB Core dump was never created, probably it is because of my environment, but I will try to get it. Server constantly eats memory, maybe some kind of memory leak? Any help is highly appreciated! Jurijs

3 25

NER and ASR
by Gholamreza Sabery 07 Nov '16

07 Nov '16

Are there any real data about number of concurrent calls, number of on-line users, NER (Network Effectiveness Ratio) and ASR (Answer Seizure Ratio) of a real VoIP system? I want this data for making comparisons. Can anyone provide a reference, article, etc..

3 4

cdr on remote POSTGRES server
by beer Ll 07 Nov '16

07 Nov '16

Hi all I have a kamailio server configured with accounting on a separate Postgres SQL server by an internal network with acc module and db_postgres module Everything works good but what can I do if I have problem on the internal network or on the postgres server ? Is it possible cache or save the data if the insert data is failed ?

2 1

fnmatch
by Slava Bendersky 07 Nov '16

07 Nov '16

Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? Slava.

3 2

kamailio + farm of Elastix + rtp direct
by Kato 06 Nov '16

06 Nov '16

Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

1 0

Kamailio DB lookup
by Gonzalo Gasca Meza 06 Nov '16

06 Nov '16

Hi Kamailions, Kamailio 4.3 will receive calls (SIP TLS) for users in this domain: *sip.parzee.io <http://sip.parzee.io>. *Example sip:gonzalo@sip.parzee.io When call is received by Kamailio I need to overwrite R-URI and transport. I should keep a mapping per user. ( gonzalo -> gonzalo58) Example. sip:gonzalo58@test.external.com;transport=tls I started looking into *dbaliases* table in LOCATION. #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif In dbaliases table I see 4 columns. Is the following configuration correct? *alias_username*: gonzalo58 *alias_domain*: test.external.com;transport=tls *username*: gonzalo *domain*: sip.parzee.io Thanks -Gonzalo

4 5

Malformed To header, or?
by Tobias 04 Nov '16

04 Nov '16

Hi! Having some issues with Kamailio 4.3 reporting a malformed header, but I cannot seem to figure out what's wrong. The problem seen is: Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/parse_addr_spec.c:678]: parse_addr_spec(): ERROR: parse_to : unexpected char [#015] in status 6: <<+123456789 <sip:123456789@127.0.0.1;tag=as4aa27bd0>> . Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/msg_parser.c:165]: get_hdr_field(): ERROR: get_hdr_field: bad to header Oct 27 13:48:56 /usr/sbin/kamailio[13765]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [To: +123456789 <si] Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: tm [t_lookup.c:1050]: t_check_msg(): ERROR: reply cannot be parsed Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/parse_addr_spec.c:678]: parse_addr_spec(): ERROR: parse_to : unexpected char [#015] in status 6: <<+123456789 <sip:123456789@127.0.0.1;tag=as4aa27bd0>> . Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/msg_parser.c:165]: get_hdr_field(): ERROR: get_hdr_field: bad to header Oct 27 13:48:56 /usr/sbin/kamailio[13765]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [To: +123456789 <si] Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [msg_translator.c:1457]: adjust_clen(): error parsing content-length Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [msg_translator.c:2214]: generate_res_buf_from_sip_res(): error while adjusting Content-Length Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [forward.c:760]: do_forward_reply(): building failed And the To header seen at this point is: To: +123456789 <sip:123456789@127.0.0.1>;tag=as24ed5606. version: kamailio 4.3.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.7.2 I obviously understand Kamailio views this To header as malformed, but what I cannot understand is why? Is it as simple as the number +123456789 is not enclosed with ", i.e. it should be sent as "+123456789"? A similar issue seen, which for some reason doesn't create an ERROR, is when Kamailio receives a reINVITE looking like this which it's supposed to proxy: To: +123456789 <sip:123456789@127.0.0.1>;tag=SD1sf9001--45026-530b969-77c7543f-530b969. which is rewritten into: To: +123456789 <sip:123456789@127*0-)'.:/)&7'...0;user=phone>;tag=SD1sf9001--45026-530b969-77c7543f-530b969. and sent out.. Thanks for your help, /Tobias

3 13

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users