12 Jun
2017
12 Jun
'17
10:25 p.m.
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
13 Jun
13 Jun
4:52 p.m.
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(a)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
_______________________________________________
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sr-users(a)lists.kamailio.org
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14 Jun
14 Jun
6:36 p.m.
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
30 Jun
30 Jun
5:22 p.m.
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(a)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
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2493
days inactive
2511
days old
3 comments
3 participants
participants (3)
-
Issam -
Issam Hambouz
-
Robert Boisvert