Hello!
I am exploring mohqueue module and have few questions:
- Could it be used to provide MOH service for SIP endpoints? I am aware that its main functionality is to queue calls (along playing a MOH for queued calls) and provides a mechanism to retrieve calls (FIFO), but I had the impression, reading several emails in this group, that it can be used in a way to provide MOH whenever a call is put on-hold -- I tried to use it with the re-INVITE received when a call is put on-hold, but got a log message stating that it works with initial INVITEs only (something like that)
- On a test setup (Kamailio 4.2.3 / 2.0.0), I was able to reach the queue / listen to the MOH but encountered a strange case where the call gets automatically removed from the queue after 4-5 seconds (can be seen removed from kamailio.mohqcalls as well)
Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): mohq_process: Processing INVITE, queue (demo) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): create_call: Added call (sip:6010@10.110.3.57;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) to queue (demo) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): first_invite_msg: Making offer for RTP link for call (<sip:6010@10.110.3.57
;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47)
Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): first_invite_msg: Sent RINGING for call (<sip:6010@10.110.3.57
;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47)
Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): send_rtp_answer: Answering RTP link for call (<sip:6010@10.110.3.57
;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47)
Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): start_stream: Starting RTP link for call (<sip:6010@10.110.3.57
;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47)
Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): send_rtp_answer: Responded to INVITE with RTP for call (< sip:6010@10.110.3.57>;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:35 ERROR: mohqueue [mohq_funcs.c:1243]: invite_cb(): invite_cb: INVITE failed for call (<sip:6010@10.110.3.57
;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47)!
Jun 12 22:58:35 DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): delete_call: Deleting call (sip:6010@10.110.3.57;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) from queue (demo)
-- route[MOH] is called first at the beginning of request_route() route[MOH] { if (has_totag()) return; if (mohq_process ()) { xlog ("mohqueue: mohq_process"); exit; } }
-- Looking at the SIP trace, all seems ok, except receiving a 404 response after BYE (that's me hanging up after few seconds from getting the above error message)
Your comments and hints are appreciated!
Thank you Issam
Issam,
mohqueue was designed to provide a way for unanswered calls to a call center or answer group to be dropped into a holding queue until it could be processed. While in queue an audio recording would be played for the caller and when a line was open to receive calls it could be directed out of the FIFO queue. It was not designed to put a call on hold, although that could be a future enhancement.
With regard to the error you identified, it appears that the INVITE did not complete. You might want to do a SIP trace and look at requests and responses to try to identify why the call failed.
Hope this helps, Bob
On Mon, Jun 12, 2017 at 4:25 PM, Issam Hambouz ihambouz@gmail.com wrote:
Hello!
I am exploring mohqueue module and have few questions:
- Could it be used to provide MOH service for SIP endpoints? I am aware
that its main functionality is to queue calls (along playing a MOH for queued calls) and provides a mechanism to retrieve calls (FIFO), but I had the impression, reading several emails in this group, that it can be used in a way to provide MOH whenever a call is put on-hold -- I tried to use it with the re-INVITE received when a call is put on-hold, but got a log message stating that it works with initial INVITEs only (something like that)
- On a test setup (Kamailio 4.2.3 / 2.0.0), I was able to reach the queue
/ listen to the MOH but encountered a strange case where the call gets automatically removed from the queue after 4-5 seconds (can be seen removed from kamailio.mohqcalls as well)
Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): mohq_process: Processing INVITE, queue (demo) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): create_call: Added call (sip:6010@10.110.3.57;tag= c97a6b26-2bfe-4271-9db1-cee38ea96c47) to queue (demo) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): first_invite_msg: Making offer for RTP link for call (< sip:6010@10.110.3.57>;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): first_invite_msg: Sent RINGING for call (sip:6010@10.110.3.57;tag= c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): send_rtp_answer: Answering RTP link for call (sip:6010@10.110.3.57;tag= c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): start_stream: Starting RTP link for call (sip:6010@10.110.3.57;tag= c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:30 - DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): send_rtp_answer: Responded to INVITE with RTP for call (< sip:6010@10.110.3.57>;tag=c97a6b26-2bfe-4271-9db1-cee38ea96c47) Jun 12 22:58:35 ERROR: mohqueue [mohq_funcs.c:1243]: invite_cb(): invite_cb: INVITE failed for call (sip:6010@10.110.3.57;tag= c97a6b26-2bfe-4271-9db1-cee38ea96c47)! Jun 12 22:58:35 DEBUG: mohqueue [mohq_funcs.c:2369]: mohq_debug(): delete_call: Deleting call (sip:6010@10.110.3.57;tag= c97a6b26-2bfe-4271-9db1-cee38ea96c47) from queue (demo)
-- route[MOH] is called first at the beginning of request_route() route[MOH] { if (has_totag()) return; if (mohq_process ()) { xlog ("mohqueue: mohq_process"); exit; } }
-- Looking at the SIP trace, all seems ok, except receiving a 404 response after BYE (that's me hanging up after few seconds from getting the above error message)
Your comments and hints are appreciated!
Thank you Issam
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello Robert,
Thank you for your response.
Point 1 is crystal clear, just needed such confirmation.
Point 2: Looking at the below trace, can't find anything wrong (except the 404 response for BYE at the end). Moreover, it looks to me that the INVITE gets positively completed, taking into account the 200 OK + MOH getting played. Yet, the mentioned error appears after 4-5 seconds, at the same moment the call gets removed from mohqcalls table .. as for the MOH, it keeps playing as expected.
Hints and comments are appreciated!
U 2017/06/14 14:56:28.132052 +0300 172.16.21.12:5060 -> 10.110.3.57:5060
INVITE sip:demo@10.110.3.57 SIP/2.0 Via: SIP/2.0/UDP 172.16.21.12:5060 ;rport;branch=z9hG4bKPjb1ac7743-b936-4074-9a5d-98e345a33f26 Max-Forwards: 70 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57 Contact: sip:6010@172.16.21.12:5060 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14866 INVITE Subject: Phone call Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH Supported: replaces Content-Type: application/sdp Content-Length: 399
v=0 o=yakama 3706430188 0 IN IP4 172.16.21.12 s=sflphone c=IN IP4 172.16.21.12 t=0 0 m=audio 32298 RTP/AVP 0 3 8 9 110 111 112 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:112 speex/32000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:32299 IN IP4 172.16.21.12
U 2017/06/14 14:56:28.132412 +0300 10.110.3.57:5060 -> 172.16.21.12:5060
SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.16.21.12:5060 ;rport=5060;branch=z9hG4bKPjb1ac7743-b936-4074-9a5d-98e345a33f26;received=172.16.21.12 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=20db48d68ad8cea70ceb54cf4988a90d.fe0d Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14866 INVITE Proxy-Authenticate: Digest realm="10.110.3.57", nonce="WUElmFlBJGysi4cnHPxRxIhhn6VcSB1i" Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0
U 2017/06/14 14:56:28.142416 +0300 172.16.21.12:5060 -> 10.110.3.57:5060
ACK sip:demo@10.110.3.57 SIP/2.0 Via: SIP/2.0/UDP 172.16.21.12:5060 ;rport;branch=z9hG4bKPjb1ac7743-b936-4074-9a5d-98e345a33f26 Max-Forwards: 70 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=20db48d68ad8cea70ceb54cf4988a90d.fe0d Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14866 ACK Content-Length: 0
U 2017/06/14 14:56:28.142768 +0300 172.16.21.12:5060 -> 10.110.3.57:5060
INVITE sip:demo@10.110.3.57 SIP/2.0 Via: SIP/2.0/UDP 172.16.21.12:5060 ;rport;branch=z9hG4bKPj63038e66-0dc3-4cce-8e07-04549f40d72b Max-Forwards: 70 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57 Contact: sip:6010@172.16.21.12:5060 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14867 INVITE Subject: Phone call Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH Supported: replaces Proxy-Authorization: Digest username="6010", realm="10.110.3.57", nonce="WUElmFlBJGysi4cnHPxRxIhhn6VcSB1i", uri="sip:demo@10.110.3.57", response="cf367b584f49117c1cf2d3a20ef629bd" Content-Type: application/sdp Content-Length: 399
v=0 o=yakama 3706430188 0 IN IP4 172.16.21.12 s=sflphone c=IN IP4 172.16.21.12 t=0 0 m=audio 32298 RTP/AVP 0 3 8 9 110 111 112 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:112 speex/32000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:32299 IN IP4 172.16.21.12
U 2017/06/14 14:56:28.148654 +0300 10.110.3.57:5060 -> 172.16.21.12:5060
SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.21.12:5060 ;rport=5060;branch=z9hG4bKPj63038e66-0dc3-4cce-8e07-04549f40d72b;received=172.16.21.12 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=2f2adbacc4317338a374a6531765d255-9497 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14867 INVITE Contact: sip:demo@10.110.3.57:5060 Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0
U 2017/06/14 14:56:28.149767 +0300 10.110.3.57:5060 -> 172.16.21.12:5060
SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.21.12:5060 ;rport=5060;branch=z9hG4bKPj63038e66-0dc3-4cce-8e07-04549f40d72b;received=172.16.21.12 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=2f2adbacc4317338a374a6531765d255-9497 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14867 INVITE Contact: sip:demo@10.110.3.57:5060 Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, PRACK Supported: 100rel Accept-Language: en Content-Type: application/sdp User-Agent: Kamailio MOH Queue v1.0 Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 166
v=0 o=- 1 1 IN IP4 10.110.3.57 s=Kamailio MOH Queue v1.0 c=IN IP4 10.110.3.57 t=0 0 a=sendrecv m=audio 49464 RTP/AVP 3 a=rtpmap:3 GSM/8000 a=nortpproxy:yes
U 2017/06/14 14:56:28.165176 +0300 172.16.21.12:5060 -> 10.110.3.57:5060
ACK sip:demo@10.110.3.57:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.21.12:5060 ;rport;branch=z9hG4bKPj69bbf40b-cc63-4699-bc05-fb41e210ac12 Max-Forwards: 70 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=2f2adbacc4317338a374a6531765d255-9497 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14867 ACK Content-Length: 0
U 2017/06/14 14:56:34.592956 +0300 172.16.21.12:5060 -> 10.110.3.57:5060
BYE sip:demo@10.110.3.57:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.21.12:5060 ;rport;branch=z9hG4bKPjd58e78ac-48c1-440c-833c-7d6b6c91038b Max-Forwards: 70 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=2f2adbacc4317338a374a6531765d255-9497 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14868 BYE Contact: sip:6010@172.16.21.12:5060 Content-Length: 0
U 2017/06/14 14:56:34.593103 +0300 10.110.3.57:5060 -> 172.16.21.12:5060
SIP/2.0 404 Not here Via: SIP/2.0/UDP 172.16.21.12:5060 ;rport=5060;branch=z9hG4bKPjd58e78ac-48c1-440c-833c-7d6b6c91038b;received=172.16.21.12 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=2f2adbacc4317338a374a6531765d255-9497 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14868 BYE Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0
Regards, Issam
Issam,
Sorry, it is vacation season and I haven't been able to get back to you on this. I recommend that in addition to analyzing the SIP trace you also enable mohqueue debugging and look at the Kamailio logs (see http://kamailio.org/docs/modules/5.0.x/modules/mohqueue.html#rpc.debug). This will tell you how mohqueue processes the messages.
Feel free to write back again but I may not be available for another week.
Bob
On Wed, Jun 14, 2017 at 12:36 PM, Issam ihambouz@gmail.com wrote:
Hello Robert,
Thank you for your response.
Point 1 is crystal clear, just needed such confirmation.
Point 2: Looking at the below trace, can't find anything wrong (except the 404 response for BYE at the end). Moreover, it looks to me that the INVITE gets positively completed, taking into account the 200 OK + MOH getting played. Yet, the mentioned error appears after 4-5 seconds, at the same moment the call gets removed from mohqcalls table .. as for the MOH, it keeps playing as expected.
Hints and comments are appreciated!
U 2017/06/14 14:56:28.132052 +0300 172.16.21.12:5060 -> 10.110.3.57:5060
INVITE sip:demo@10.110.3.57 SIP/2.0 Via: SIP/2.0/UDP 172.16.21.12:5060;rport;branch=z9hG4bKPjb1ac7743-b936- 4074-9a5d-98e345a33f26 Max-Forwards: 70 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57 Contact: sip:6010@172.16.21.12:5060 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14866 INVITE Subject: Phone call Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH Supported: replaces Content-Type: application/sdp Content-Length: 399
v=0 o=yakama 3706430188 0 IN IP4 172.16.21.12 s=sflphone c=IN IP4 172.16.21.12 t=0 0 m=audio 32298 RTP/AVP 0 3 8 9 110 111 112 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:112 speex/32000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:32299 IN IP4 172.16.21.12
U 2017/06/14 14:56:28.132412 +0300 10.110.3.57:5060 -> 172.16.21.12:5060
SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.16.21.12:5060;rport=5060; branch=z9hG4bKPjb1ac7743-b936-4074-9a5d-98e345a33f26;received=172.16.21.12 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=20db48d68ad8cea70ceb54cf4988a90d.fe0d Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14866 INVITE Proxy-Authenticate: Digest realm="10.110.3.57", nonce=" WUElmFlBJGysi4cnHPxRxIhhn6VcSB1i" Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0
U 2017/06/14 14:56:28.142416 +0300 172.16.21.12:5060 -> 10.110.3.57:5060
ACK sip:demo@10.110.3.57 SIP/2.0 Via: SIP/2.0/UDP 172.16.21.12:5060;rport;branch=z9hG4bKPjb1ac7743-b936- 4074-9a5d-98e345a33f26 Max-Forwards: 70 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=20db48d68ad8cea70ceb54cf4988a90d.fe0d Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14866 ACK Content-Length: 0
U 2017/06/14 14:56:28.142768 +0300 172.16.21.12:5060 -> 10.110.3.57:5060
INVITE sip:demo@10.110.3.57 SIP/2.0 Via: SIP/2.0/UDP 172.16.21.12:5060;rport;branch=z9hG4bKPj63038e66-0dc3- 4cce-8e07-04549f40d72b Max-Forwards: 70 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57 Contact: sip:6010@172.16.21.12:5060 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14867 INVITE Subject: Phone call Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS, MESSAGE, PUBLISH Supported: replaces Proxy-Authorization: Digest username="6010", realm="10.110.3.57", nonce=" WUElmFlBJGysi4cnHPxRxIhhn6VcSB1i", uri="sip:demo@10.110.3.57", response=" cf367b584f49117c1cf2d3a20ef629bd" Content-Type: application/sdp Content-Length: 399
v=0 o=yakama 3706430188 0 IN IP4 172.16.21.12 s=sflphone c=IN IP4 172.16.21.12 t=0 0 m=audio 32298 RTP/AVP 0 3 8 9 110 111 112 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:112 speex/32000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:32299 IN IP4 172.16.21.12
U 2017/06/14 14:56:28.148654 +0300 10.110.3.57:5060 -> 172.16.21.12:5060
SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.21.12:5060;rport=5060; branch=z9hG4bKPj63038e66-0dc3-4cce-8e07-04549f40d72b;received=172.16.21.12 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=2f2adbacc4317338a374a6531765d255-9497 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14867 INVITE Contact: sip:demo@10.110.3.57:5060 Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0
U 2017/06/14 14:56:28.149767 +0300 10.110.3.57:5060 -> 172.16.21.12:5060
SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.21.12:5060;rport=5060; branch=z9hG4bKPj63038e66-0dc3-4cce-8e07-04549f40d72b;received=172.16.21.12 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=2f2adbacc4317338a374a6531765d255-9497 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14867 INVITE Contact: sip:demo@10.110.3.57:5060 Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, PRACK Supported: 100rel Accept-Language: en Content-Type: application/sdp User-Agent: Kamailio MOH Queue v1.0 Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 166
v=0 o=- 1 1 IN IP4 10.110.3.57 s=Kamailio MOH Queue v1.0 c=IN IP4 10.110.3.57 t=0 0 a=sendrecv m=audio 49464 RTP/AVP 3 a=rtpmap:3 GSM/8000 a=nortpproxy:yes
U 2017/06/14 14:56:28.165176 +0300 172.16.21.12:5060 -> 10.110.3.57:5060
ACK sip:demo@10.110.3.57:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.21.12:5060;rport;branch=z9hG4bKPj69bbf40b-cc63- 4699-bc05-fb41e210ac12 Max-Forwards: 70 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=2f2adbacc4317338a374a6531765d255-9497 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14867 ACK Content-Length: 0
U 2017/06/14 14:56:34.592956 +0300 172.16.21.12:5060 -> 10.110.3.57:5060
BYE sip:demo@10.110.3.57:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.21.12:5060;rport;branch=z9hG4bKPjd58e78ac-48c1- 440c-833c-7d6b6c91038b Max-Forwards: 70 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=2f2adbacc4317338a374a6531765d255-9497 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14868 BYE Contact: sip:6010@172.16.21.12:5060 Content-Length: 0
U 2017/06/14 14:56:34.593103 +0300 10.110.3.57:5060 -> 172.16.21.12:5060
SIP/2.0 404 Not here Via: SIP/2.0/UDP 172.16.21.12:5060;rport=5060; branch=z9hG4bKPjd58e78ac-48c1-440c-833c-7d6b6c91038b;received=172.16.21.12 From: sip:6010@10.110.3.57;tag=8bfd31ff-59ad-4efd-8ae5-47fde1862ef1 To: sip:demo@10.110.3.57;tag=2f2adbacc4317338a374a6531765d255-9497 Call-ID: 2116d060-8211-4f5c-beba-4a7d6c4fd08f CSeq: 14868 BYE Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0
Regards, Issam
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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Issam -
Issam Hambouz
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Robert Boisvert