Hi all,
I am still stuck with the ACK message not being forwarded by the originating PCSCF. Any advice would be great.
Thanks, Serhat
On 24 October 2016 at 21:00, Serhat Guler srtguler@gmail.com wrote:
Hi Daniel,
I am using only record_route() without any parameters. I do not have a proper computer atm to draw the network diagram, but I can tell you shortly about the network setup.
I have only enabled websockets for the pcscf to allow ws and wss connections. In that case there is a ws connection that uses UDP protocol. This is the ACK to complete the session setup.
the sipml5 client is configured as follows: WebSocket Server URL: ws://192.168.0.11:880 SIP outbound Proxy URL: udp://192.168.0.11:4060
Mercuro IMS client: uses UDP port as well: 4060
The call is made from sipml5 client. The Mercuro phone rings, and when I reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from sipml5 doesn't pass the PCSCF as I explained in the previous message.
A part of PCSCF cfg file:
# Check for Subsequent requests: if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if ($route_uri =~ "sip:mo@.*") { setflag(FLT_MO); } if(!isdsturiset()) { handle_ruri_alias(); } # RTP-Relay, if necessary route(RTPPROXY); t_relay(); } else { 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 { xlog("L_INFO", "ACK without matching transaction ...ignore and discard!!!!!\n"); # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; }
Cheers, Serhat
On 24 October 2016 at 20:18, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
I haven't noticed the log files, it's ok.
From the Route header, I see that there is a proxy that uses WS:
Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy 1nCMEI1mR0RztrB;did=e82.0c3 That is the address of the next hop and typically a proxy doesn't use websocket connection to another proxy. Can you show a diagram with the sip server nodes in your network and what protocols are used between them?
Are you simply use record_route() function, or some other function or different parameters to it?
Cheers, Daniel
On 24/10/16 12:18, Serhat Guler wrote:
Hi Daniel,
Thanks for your reply. I actually attached a log file with debug level 3, consisting ACK related messages. If you would like to see more logs, I'll send a new log file in the evening.
Cheers, Serhat
On 24 October 2016 at 12:13, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
can you get all the log messages for ACK but with debug=3 in the kamailio.cfg?
Cheers, Daniel
On 23/10/16 22:04, Serhat Guler wrote:
Hello,
I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says:
8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed
I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5.
The ACK package:
ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9 hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"sip:bob@net1.test;tag=GxzKy1nCMEI1mR0RztrB To: sip:alice@net1.test;tag=18823 Contact: "Bob"sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2c all=no;transport=ws;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp Max-Forwards: 69 Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy 1nCMEI1mR0RztrB;did=e82.0c3 Route: sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz trB;did=e82.0c3 Route: sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.f062 Route: sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.f062 Route: sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.1c3 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom
Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached.
Thanks in advance, Serhat
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cg i-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
Hi Serhat,
I am not sure how is the setup of your network, but you should remove the outbound proxy setting from sipml5 (SIP outbound Proxy URL: udp:// 192.168.0.11:4060).
Test it and let us know.
Regards,
On Sat, Oct 29, 2016 at 9:38 PM, Serhat Guler srtguler@gmail.com wrote:
Hi all,
I am still stuck with the ACK message not being forwarded by the originating PCSCF. Any advice would be great.
Thanks, Serhat
On 24 October 2016 at 21:00, Serhat Guler srtguler@gmail.com wrote:
Hi Daniel,
I am using only record_route() without any parameters. I do not have a proper computer atm to draw the network diagram, but I can tell you shortly about the network setup.
I have only enabled websockets for the pcscf to allow ws and wss connections. In that case there is a ws connection that uses UDP protocol. This is the ACK to complete the session setup.
the sipml5 client is configured as follows: WebSocket Server URL: ws://192.168.0.11:880 SIP outbound Proxy URL: udp://192.168.0.11:4060
Mercuro IMS client: uses UDP port as well: 4060
The call is made from sipml5 client. The Mercuro phone rings, and when I reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from sipml5 doesn't pass the PCSCF as I explained in the previous message.
A part of PCSCF cfg file:
# Check for Subsequent requests: if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if ($route_uri =~ "sip:mo@.*") { setflag(FLT_MO); } if(!isdsturiset()) { handle_ruri_alias(); } # RTP-Relay, if necessary route(RTPPROXY); t_relay(); } else { 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 { xlog("L_INFO", "ACK without matching transaction ...ignore and discard!!!!!\n"); # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; }
Cheers, Serhat
On 24 October 2016 at 20:18, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
I haven't noticed the log files, it's ok.
From the Route header, I see that there is a proxy that uses WS:
Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy 1nCMEI1mR0RztrB;did=e82.0c3 That is the address of the next hop and typically a proxy doesn't use websocket connection to another proxy. Can you show a diagram with the sip server nodes in your network and what protocols are used between them?
Are you simply use record_route() function, or some other function or different parameters to it?
Cheers, Daniel
On 24/10/16 12:18, Serhat Guler wrote:
Hi Daniel,
Thanks for your reply. I actually attached a log file with debug level 3, consisting ACK related messages. If you would like to see more logs, I'll send a new log file in the evening.
Cheers, Serhat
On 24 October 2016 at 12:13, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
Hello,
can you get all the log messages for ACK but with debug=3 in the kamailio.cfg?
Cheers, Daniel
On 23/10/16 22:04, Serhat Guler wrote:
Hello,
I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says:
8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed
I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5.
The ACK package:
ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9 hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"sip:bob@net1.test;tag=GxzKy1nCMEI1mR0RztrB To: sip:alice@net1.test;tag=18823 Contact: "Bob"sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2c all=no;transport=ws;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp Max-Forwards: 69 Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy 1nCMEI1mR0RztrB;did=e82.0c3 Route: sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz trB;did=e82.0c3 Route: sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.f062 Route: sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.f062 Route: sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.1c3 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom
Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached.
Thanks in advance, Serhat
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cg i-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi Alberto,
Removing the outbound proxy solved the problem. Thanks for your help.
Cheers, Serhat
On 30 October 2016 at 10:52, Alberto Llamas albertollamaso@gmail.com wrote:
Hi Serhat,
I am not sure how is the setup of your network, but you should remove the outbound proxy setting from sipml5 (SIP outbound Proxy URL: udp:// 192.168.0.11:4060).
Test it and let us know.
Regards,
On Sat, Oct 29, 2016 at 9:38 PM, Serhat Guler srtguler@gmail.com wrote:
Hi all,
I am still stuck with the ACK message not being forwarded by the originating PCSCF. Any advice would be great.
Thanks, Serhat
On 24 October 2016 at 21:00, Serhat Guler srtguler@gmail.com wrote:
Hi Daniel,
I am using only record_route() without any parameters. I do not have a proper computer atm to draw the network diagram, but I can tell you shortly about the network setup.
I have only enabled websockets for the pcscf to allow ws and wss connections. In that case there is a ws connection that uses UDP protocol. This is the ACK to complete the session setup.
the sipml5 client is configured as follows: WebSocket Server URL: ws://192.168.0.11:880 SIP outbound Proxy URL: udp://192.168.0.11:4060
Mercuro IMS client: uses UDP port as well: 4060
The call is made from sipml5 client. The Mercuro phone rings, and when I reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from sipml5 doesn't pass the PCSCF as I explained in the previous message.
A part of PCSCF cfg file:
# Check for Subsequent requests: if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if ($route_uri =~ "sip:mo@.*") { setflag(FLT_MO); } if(!isdsturiset()) { handle_ruri_alias(); } # RTP-Relay, if necessary route(RTPPROXY); t_relay(); } else { 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 { xlog("L_INFO", "ACK without matching transaction ...ignore and discard!!!!!\n"); # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; }
Cheers, Serhat
On 24 October 2016 at 20:18, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
Hello,
I haven't noticed the log files, it's ok.
From the Route header, I see that there is a proxy that uses WS:
Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy 1nCMEI1mR0RztrB;did=e82.0c3 That is the address of the next hop and typically a proxy doesn't use websocket connection to another proxy. Can you show a diagram with the sip server nodes in your network and what protocols are used between them?
Are you simply use record_route() function, or some other function or different parameters to it?
Cheers, Daniel
On 24/10/16 12:18, Serhat Guler wrote:
Hi Daniel,
Thanks for your reply. I actually attached a log file with debug level 3, consisting ACK related messages. If you would like to see more logs, I'll send a new log file in the evening.
Cheers, Serhat
On 24 October 2016 at 12:13, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
can you get all the log messages for ACK but with debug=3 in the kamailio.cfg?
Cheers, Daniel
On 23/10/16 22:04, Serhat Guler wrote:
Hello,
I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says:
8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed
I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5.
The ACK package:
ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9 hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"sip:bob@net1.test;tag=GxzKy1nCMEI1mR0RztrB To: sip:alice@net1.test;tag=18823 Contact: "Bob"sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2c all=no;transport=ws;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp Max-Forwards: 69 Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy 1nCMEI1mR0RztrB;did=e82.0c3 Route: sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz trB;did=e82.0c3 Route: sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.f062 Route: sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.f062 Route: sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.1c3 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom
Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached.
Thanks in advance, Serhat
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cg i-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Alberto Llamas Phone: +1-786-805-6003 Telecommunications Engineer Digium Certified Asterisk Professional (dCap)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Excellent !
On Sun, Oct 30, 2016 at 8:33 PM, Serhat Guler srtguler@gmail.com wrote:
Hi Alberto,
Removing the outbound proxy solved the problem. Thanks for your help.
Cheers, Serhat
On 30 October 2016 at 10:52, Alberto Llamas albertollamaso@gmail.com wrote:
Hi Serhat,
I am not sure how is the setup of your network, but you should remove the outbound proxy setting from sipml5 (SIP outbound Proxy URL: udp:// 192.168.0.11:4060).
Test it and let us know.
Regards,
On Sat, Oct 29, 2016 at 9:38 PM, Serhat Guler srtguler@gmail.com wrote:
Hi all,
I am still stuck with the ACK message not being forwarded by the originating PCSCF. Any advice would be great.
Thanks, Serhat
On 24 October 2016 at 21:00, Serhat Guler srtguler@gmail.com wrote:
Hi Daniel,
I am using only record_route() without any parameters. I do not have a proper computer atm to draw the network diagram, but I can tell you shortly about the network setup.
I have only enabled websockets for the pcscf to allow ws and wss connections. In that case there is a ws connection that uses UDP protocol. This is the ACK to complete the session setup.
the sipml5 client is configured as follows: WebSocket Server URL: ws://192.168.0.11:880 SIP outbound Proxy URL: udp://192.168.0.11:4060
Mercuro IMS client: uses UDP port as well: 4060
The call is made from sipml5 client. The Mercuro phone rings, and when I reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from sipml5 doesn't pass the PCSCF as I explained in the previous message.
A part of PCSCF cfg file:
# Check for Subsequent requests: if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if ($route_uri =~ "sip:mo@.*") { setflag(FLT_MO); } if(!isdsturiset()) { handle_ruri_alias(); } # RTP-Relay, if necessary route(RTPPROXY); t_relay(); } else { 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 { xlog("L_INFO", "ACK without matching transaction... ignore and discard!!!!!\n"); # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; }
Cheers, Serhat
On 24 October 2016 at 20:18, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
I haven't noticed the log files, it's ok.
From the Route header, I see that there is a proxy that uses WS:
Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy 1nCMEI1mR0RztrB;did=e82.0c3 That is the address of the next hop and typically a proxy doesn't use websocket connection to another proxy. Can you show a diagram with the sip server nodes in your network and what protocols are used between them?
Are you simply use record_route() function, or some other function or different parameters to it?
Cheers, Daniel
On 24/10/16 12:18, Serhat Guler wrote:
Hi Daniel,
Thanks for your reply. I actually attached a log file with debug level 3, consisting ACK related messages. If you would like to see more logs, I'll send a new log file in the evening.
Cheers, Serhat
On 24 October 2016 at 12:13, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
Hello,
can you get all the log messages for ACK but with debug=3 in the kamailio.cfg?
Cheers, Daniel
On 23/10/16 22:04, Serhat Guler wrote:
Hello,
I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says:
8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed
I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5.
The ACK package:
ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9 hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"sip:bob@net1.test;tag=GxzKy1nCMEI1mR0RztrB To: sip:alice@net1.test;tag=18823 Contact: "Bob"sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2c all=no;transport=ws;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp Max-Forwards: 69 Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy 1nCMEI1mR0RztrB;did=e82.0c3 Route: sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz trB;did=e82.0c3 Route: sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.f062 Route: sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.f062 Route: sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di d=e82.1c3 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom
Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached.
Thanks in advance, Serhat
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cg i-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Alberto Llamas Phone: +1-786-805-6003 Telecommunications Engineer Digium Certified Asterisk Professional (dCap)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Alberto Llamas -
Serhat Guler