23 Feb
2004
23 Feb
'04
4:38 p.m.
Hi all,
I have been trying to solve this problem for a couple of hours, but i could'n. Could
somebody help me?
I have configure a voice mail system (http://iptel.org/~faqomatic/fom-serve/cache/92.html)
and I have a problem when it answer the call; it doesn't stop recording when I hang on
the phone and never sends the mail. this is the sems debug:
------------------------------------------------------------------------------------------------------------------------------------------------
(815) DEBUG: onSessionStart (AnswerMachine.cpp:170): starting recording
message...(/tmp/2870083197(a)200.69.228.52.wav)
(815) DEBUG: onSessionStart (AnswerMachine.cpp:173): ... finished recording message
(815) DEBUG: sendToFIFO (AmRequest.cpp:199): msg=<:t_uac_dlg:0000032F15FB7483
BYE
sip:100@200.69.228.52:5060;user=phone;transport=udp
sip:200@200.69.228.49;ftag=450631392;lr=on
From: <sip:200@200.69.228.49;user=phone>;tag=0000032F41C67EA6
To: <sip:100@200.69.228.49;user=phone>;tag=450631392
CSeq: 2 BYE
Call-ID: 2870083197(a)200.69.228.52
Contact: <sip:200@200.69.228.49:5090
Route: <sip:200@200.69.228.49;ftag=450631392;lr=on
.
.
(815) DEBUG: write_to_fifo (AmRequest.cpp:708):
Write to fifo: completed
---------------------------------------------------------------------------------------------------
it never never gets the:
DEBUG: getReturnCode (AmRequest.cpp:421): response from Ser: 200 fifo_t_reply succeeded
and sends the mail. When I call the voice mail with an ATA pointing to port 5090 (SER
voice mail) it works ok and sends the mail.
The system is configured with 2 SER, one for routing and the other for voice mail.
Find attached the configuration files.
Thank you in advance
Juan Priotti
23 Feb
23 Feb
5:06 p.m.
New subject: [Serusers] voice mail doesn't stop redording and never sends voicemail - need help
Have you seen the BYE on the net? And a reply to it? -jiri
At 10:38 PM 2/23/2004, Juan wrote:
Hi all,
I have been trying to solve this problem for a couple of hours, but i could'n. Could
somebody help me?
I have configure a voice mail system
(<http://iptel.org/~faqomatic/fom-serve/cache/92.html>http://iptel.org/~faqomatic/fom-serve/cache/92.html)
and I have a problem when it answer the call; it doesn't stop recording when I hang on
the phone and never sends the mail. this is the sems debug:
------------------------------------------------------------------------------------------------------------------------------------------------
(815) DEBUG: onSessionStart (AnswerMachine.cpp:170): starting recording
message...(/tmp/2870083197(a)200.69.228.52.wav)
(815) DEBUG: onSessionStart (AnswerMachine.cpp:173): ... finished recording message
(815) DEBUG: sendToFIFO (AmRequest.cpp:199): msg=<:t_uac_dlg:0000032F15FB7483
BYE
sip:100@200.69.228.52:5060;user=phone;transport=udp
sip:200@200.69.228.49;ftag=450631392;lr=on
From: <sip:200@200.69.228.49;user=phone>;tag=0000032F41C67EA6
To: <sip:100@200.69.228.49;user=phone>;tag=450631392
CSeq: 2 BYE
Call-ID: <mailto:2870083197@200.69.228.52>2870083197@200.69.228.52
Contact: <sip:200@200.69.228.49:5090
Route: <sip:200@200.69.228.49;ftag=450631392;lr=on .
.
(815) DEBUG: write_to_fifo
(AmRequest.cpp:708): Write to fifo: completed
---------------------------------------------------------------------------------------------------
it never never gets the:
DEBUG: getReturnCode (AmRequest.cpp:421): response from Ser: 200 fifo_t_reply succeeded
and sends the mail. When I call the voice mail with an ATA pointing to port 5090 (SER
voice mail) it works ok and sends the mail.
The system is configured with 2 SER, one for routing and the other for voice mail.
Find attached the configuration files.
Thank you in advance
Juan Priotti
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
5:47 p.m.
I can see the bye and the reply to it. This is what I see on the net:
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
INVITE sip:200@200.69.228.49;user=phone SIP/2.0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;user=phone>
;tag=3788800848..To: <sip:200@200.69.228.49;user=phone>..Call-ID:
3273316177@200.69.228.52..CSeq: 1 INVITE..Contact: <sip:100
@200.69.228.52:5060;user=phone;transport=udp>..User-Agent: Cisco ATA
v2.15 ata18x (020927a)..Expires: 300..Content-Length: 248
..Content-Type: application/sdp....v=0..o=100 13453 13453 IN IP4
200.69.228.52..s=ATA186 Call..c=IN IP4 200.69.228.52..t=0 0..m
=audio 16384 RTP/AVP 0 4 8 101..a=rtpmap:0 PCMU/8000/1..a=rtpmap:4
G723/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:101 telephone-
event/8000..a=fmtp:101 0-15..
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;user=pho
ne>;tag=3788800848..To: <sip:200@200.69.228.49;user=phone>..Call-ID:
3273316177@200.69.228.52..CSeq: 1 INVITE..Server: Sip EXp
ress router (0.8.12 (i386/freebsd))..Content-Length: 0..Warning: 392
200.69.228.49:5060 "Noisy feedback tells: pid=921 req_src
_ip=200.69.228.52 req_src_port=5060
in_uri=sip:200@200.69.228.49;user=phone
out_uri=sip:200@200.69.228.39:5060;user=phone;trans
port=udp via_cnt==1"....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
INVITE sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag
=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.49;branch=z9hG4bKe8cb.d06324d3.0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <
sip:100@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>..Call-ID: 3273316177@200.69.228.52..C
Seq: 1 INVITE..Contact:
<sip:100@200.69.228.52:5060;user=phone;transport=udp>..User-Agent: Cisco ATA
v2.15 ata18x (020927a)..
Expires: 300..Content-Length: 248..Content-Type:
application/sdp....v=0..o=100 13453 13453 IN IP4 200.69.228.52..s=ATA186
Call.
.c=IN IP4 200.69.228.52..t=0 0..m=audio 16384 RTP/AVP 0 4 8
101..a=rtpmap:0 PCMU/8000/1..a=rtpmap:4 G723/8000/1..a=rtpmap:8 PCM
A/8000/1..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..
#
U 200.69.228.39:5060 -> 200.69.228.49:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
200.69.228.49;branch=z9hG4bKe8cb.d06324d3.0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <
sip:100@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=2822163365..Call-ID: 3273316177@2
00.69.228.52..CSeq: 1 INVITE..Server: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.39:5060 -> 200.69.228.49:5060
SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
200.69.228.49;branch=z9hG4bKe8cb.d06324d3.0..Via: SIP/2.0/UDP
200.69.228.52:5060..From:
<sip:100@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=2822163365..Call-ID: 3273316177@
200.69.228.52..CSeq: 1 INVITE..Server: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 200.69.228.52:5060..From:
<sip:100@200.69.228.49;user=phone>;tag=3788800848..To: <sip:2
00@200.69.228.49;user=phone>;tag=2822163365..Call-ID:
3273316177@200.69.228.52..CSeq: 1 INVITE..Server: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
CANCEL sip:200@200.69.228.39:5060;user=phone;transport=udp SIP/2.0..Via:
SIP/2.0/UDP 200.69.228.49;branch=z9hG4bKe8cb.d06324d3.
0..From: <sip:100@200.69.228.49;user=phone>;tag=3788800848..Call-ID:
3273316177@200.69.228.52..To: <sip:200@200.69.228.49;use
r=phone>..CSeq: 1 CANCEL..User-Agent: Sip EXpress router(0.8.12
(i386/freebsd))..Content-Length: 0....
#
U 200.69.228.39:5060 -> 200.69.228.49:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP
200.69.228.49;branch=z9hG4bKe8cb.d06324d3.0..From:
<sip:100@200.69.228.49;user=phone>;tag=378
8800848..To: <sip:200@200.69.228.49;user=phone>;tag=2822163365..Call-ID:
3273316177@200.69.228.52..CSeq: 1 CANCEL..Server: Cis
co ATA v2.15 ata18x (020927a)..Content-Length: 0....
#
U 200.69.228.39:5060 -> 200.69.228.49:5060
SIP/2.0 487 Request Cancelled..Via: SIP/2.0/UDP
200.69.228.49;branch=z9hG4bKe8cb.d06324d3.0..Via: SIP/2.0/UDP
200.69.228.52:506
0..From: <sip:100@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=2822163365..Call-ID: 3
273316177@200.69.228.52..CSeq: 1 INVITE..Server: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
ACK sip:200@200.69.228.39:5060;user=phone;transport=udp SIP/2.0..Via:
SIP/2.0/UDP 200.69.228.49;branch=z9hG4bKe8cb.d06324d3.0..
From: <sip:100@200.69.228.49;user=phone>;tag=3788800848..Call-ID:
3273316177@200.69.228.52..To: <sip:200@200.69.228.49;user=p
hone>;tag=2822163365..CSeq: 1 ACK..User-Agent: Sip EXpress router(0.8.12
(i386/freebsd))..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 180 ringing..Record-Route:
<sip:200@200.69.228.49;ftag=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <
sip:100@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID: 327331
6177@200.69.228.52..CSeq: 1 INVITE..Contact:
<sip:200@200.69.228.49:5090>..Server: Sip EXpress router (0.8.12
(i386/freebsd))..
Content-Length: 0..Warning: 392 200.69.228.49:5090 "Noisy feedback tells:
pid=956 req_src_ip=200.69.228.49 req_src_port=5060 i
n_uri=sip:200@200.69.228.49:5090;user=phone
out_uri=sip:200@200.69.228.49:5090;user=phone via_cnt==0"....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 200 OK..Record-Route:
<sip:200@200.69.228.49;ftag=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 INVITE..Contact:
<sip:200@200.69.228.49:5090>..Content-Type: application/sdp..Server: Sip
EXpress router
(0.8.12 (i386/freebsd))..Content-Length: 131..Warning: 392
200.69.228.49:5090 "Noisy feedback tells: pid=956 req_src_ip=200.6
9.228.49 req_src_port=5060 in_uri=sip:200@200.69.228.49:5090;user=phone
out_uri=sip:200@200.69.228.49:5090;user=phone via_cnt==
0"....v=0..o=username 0 0 IN IP4 200.69.228.49..s=session..c=IN IP4
200.69.228.49..t=0 0..m=audio 58516 RTP/AVP 0..a=rtpmap:0 P
CMU/8000..
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
ACK sip:200@200.69.228.49 SIP/2.0..Route:
<sip:200@200.69.228.49:5090>..Via: SIP/2.0/UDP 200.69.228.52:5060..From:
<sip:100@20
0.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177(a)200.6
9.228.52..CSeq: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
ACK sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag=37
88800848;lr=on>..Via: SIP/2.0/UDP 200.69.228.49;branch=0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;us
er=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq
: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x (020927a)..Content-Length:
0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 200 OK..Record-Route:
<sip:200@200.69.228.49;ftag=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 INVITE..Contact:
<sip:200@200.69.228.49:5090>..Content-Type: application/sdp..Server: Sip
EXpress router
(0.8.12 (i386/freebsd))..Content-Length: 131..Warning: 392
200.69.228.49:5090 "Noisy feedback tells: pid=956 req_src_ip=200.6
9.228.49 req_src_port=5060 in_uri=sip:200@200.69.228.49:5090;user=phone
out_uri=sip:200@200.69.228.49:5090;user=phone via_cnt==
0"....v=0..o=username 0 0 IN IP4 200.69.228.49..s=session..c=IN IP4
200.69.228.49..t=0 0..m=audio 58516 RTP/AVP 0..a=rtpmap:0 P
CMU/8000..
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
ACK sip:200@200.69.228.49:5060 SIP/2.0..Route:
<sip:200@200.69.228.49:5090>..Via: SIP/2.0/UDP 200.69.228.52:5060..From:
<sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
ACK sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag=37
88800848;lr=on>..Via: SIP/2.0/UDP 200.69.228.49;branch=0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;us
er=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq
: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x (020927a)..Content-Length:
0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 200 OK..Record-Route:
<sip:200@200.69.228.49;ftag=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 INVITE..Contact:
<sip:200@200.69.228.49:5090>..Content-Type: application/sdp..Server: Sip
EXpress router
(0.8.12 (i386/freebsd))..Content-Length: 131..Warning: 392
200.69.228.49:5090 "Noisy feedback tells: pid=956 req_src_ip=200.6
9.228.49 req_src_port=5060 in_uri=sip:200@200.69.228.49:5090;user=phone
out_uri=sip:200@200.69.228.49:5090;user=phone via_cnt==
0"....v=0..o=username 0 0 IN IP4 200.69.228.49..s=session..c=IN IP4
200.69.228.49..t=0 0..m=audio 58516 RTP/AVP 0..a=rtpmap:0 P
CMU/8000..
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
ACK sip:200@200.69.228.49:5060 SIP/2.0..Route:
<sip:200@200.69.228.49:5090>..Via: SIP/2.0/UDP 200.69.228.52:5060..From:
<sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
ACK sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag=37
88800848;lr=on>..Via: SIP/2.0/UDP 200.69.228.49;branch=0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;us
er=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq
: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x (020927a)..Content-Length:
0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 200 OK..Record-Route:
<sip:200@200.69.228.49;ftag=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 INVITE..Contact:
<sip:200@200.69.228.49:5090>..Content-Type: application/sdp..Server: Sip
EXpress router
(0.8.12 (i386/freebsd))..Content-Length: 131..Warning: 392
200.69.228.49:5090 "Noisy feedback tells: pid=956 req_src_ip=200.6
9.228.49 req_src_port=5060 in_uri=sip:200@200.69.228.49:5090;user=phone
out_uri=sip:200@200.69.228.49:5090;user=phone via_cnt==
0"....v=0..o=username 0 0 IN IP4 200.69.228.49..s=session..c=IN IP4
200.69.228.49..t=0 0..m=audio 58516 RTP/AVP 0..a=rtpmap:0 P
CMU/8000..
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
ACK sip:200@200.69.228.49:5060 SIP/2.0..Route:
<sip:200@200.69.228.49:5090>..Via: SIP/2.0/UDP 200.69.228.52:5060..From:
<sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
ACK sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag=37
88800848;lr=on>..Via: SIP/2.0/UDP 200.69.228.49;branch=0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;us
er=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq
: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x (020927a)..Content-Length:
0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 200 OK..Record-Route:
<sip:200@200.69.228.49;ftag=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 INVITE..Contact:
<sip:200@200.69.228.49:5090>..Content-Type: application/sdp..Server: Sip
EXpress router
(0.8.12 (i386/freebsd))..Content-Length: 131..Warning: 392
200.69.228.49:5090 "Noisy feedback tells: pid=956 req_src_ip=200.6
9.228.49 req_src_port=5060 in_uri=sip:200@200.69.228.49:5090;user=phone
out_uri=sip:200@200.69.228.49:5090;user=phone via_cnt==
0"....v=0..o=username 0 0 IN IP4 200.69.228.49..s=session..c=IN IP4
200.69.228.49..t=0 0..m=audio 58516 RTP/AVP 0..a=rtpmap:0 P
CMU/8000..
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
ACK sip:200@200.69.228.49:5060 SIP/2.0..Route:
<sip:200@200.69.228.49:5090>..Via: SIP/2.0/UDP 200.69.228.52:5060..From:
<sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
ACK sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag=37
88800848;lr=on>..Via: SIP/2.0/UDP 200.69.228.49;branch=0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;us
er=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq
: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x (020927a)..Content-Length:
0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 200 OK..Record-Route:
<sip:200@200.69.228.49;ftag=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 INVITE..Contact:
<sip:200@200.69.228.49:5090>..Content-Type: application/sdp..Server: Sip
EXpress router
(0.8.12 (i386/freebsd))..Content-Length: 131..Warning: 392
200.69.228.49:5090 "Noisy feedback tells: pid=956 req_src_ip=200.6
9.228.49 req_src_port=5060 in_uri=sip:200@200.69.228.49:5090;user=phone
out_uri=sip:200@200.69.228.49:5090;user=phone via_cnt==
0"....v=0..o=username 0 0 IN IP4 200.69.228.49..s=session..c=IN IP4
200.69.228.49..t=0 0..m=audio 58516 RTP/AVP 0..a=rtpmap:0 P
CMU/8000..
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
ACK sip:200@200.69.228.49:5060 SIP/2.0..Route:
<sip:200@200.69.228.49:5090>..Via: SIP/2.0/UDP 200.69.228.52:5060..From:
<sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
ACK sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag=37
88800848;lr=on>..Via: SIP/2.0/UDP 200.69.228.49;branch=0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;us
er=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq
: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x (020927a)..Content-Length:
0....
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
BYE sip:200@200.69.228.49 SIP/2.0..Route:
<sip:200@200.69.228.49:5090>..Via: SIP/2.0/UDP 200.69.228.52:5060..From:
<sip:100@20
0.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177(a)200.6
9.228.52..CSeq: 2 BYE..User-Agent: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
BYE sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag=37
88800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.49;branch=z9hG4bKb8cb.6304f3c2.0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip
:100@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID: 327331617
7@200.69.228.52..CSeq: 2 BYE..User-Agent: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.39:5060 -> 200.69.228.49:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP
200.69.228.49;branch=z9hG4bKb8cb.6304f3c2.0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:
100@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID: 3273316177
@200.69.228.52..CSeq: 2 BYE..Server: Cisco ATA v2.15 ata18x
(020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 200.69.228.52:5060..From:
<sip:100@200.69.228.49;user=phone>;tag=3788800848..To: <sip:200@20
0.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq: 2 BYE..Server: Cisco ATA v2.15 ata18x (
020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 200 OK..Record-Route:
<sip:200@200.69.228.49;ftag=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 INVITE..Contact:
<sip:200@200.69.228.49:5090>..Content-Type: application/sdp..Server: Sip
EXpress router
(0.8.12 (i386/freebsd))..Content-Length: 131..Warning: 392
200.69.228.49:5090 "Noisy feedback tells: pid=956 req_src_ip=200.6
9.228.49 req_src_port=5060 in_uri=sip:200@200.69.228.49:5090;user=phone
out_uri=sip:200@200.69.228.49:5090;user=phone via_cnt==
0"....v=0..o=username 0 0 IN IP4 200.69.228.49..s=session..c=IN IP4
200.69.228.49..t=0 0..m=audio 58516 RTP/AVP 0..a=rtpmap:0 P
CMU/8000..
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
ACK sip:200@200.69.228.49:5060 SIP/2.0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;user=phone>;tag=3788
800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq: 1 ACK..User-Agen
t: Cisco ATA v2.15 ata18x (020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
ACK sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag=37
88800848;lr=on>..Via: SIP/2.0/UDP 200.69.228.49;branch=0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;us
er=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq
: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x (020927a)..Content-Length:
0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 200 OK..Record-Route:
<sip:200@200.69.228.49;ftag=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 INVITE..Contact:
<sip:200@200.69.228.49:5090>..Content-Type: application/sdp..Server: Sip
EXpress router
(0.8.12 (i386/freebsd))..Content-Length: 131..Warning: 392
200.69.228.49:5090 "Noisy feedback tells: pid=956 req_src_ip=200.6
9.228.49 req_src_port=5060 in_uri=sip:200@200.69.228.49:5090;user=phone
out_uri=sip:200@200.69.228.49:5090;user=phone via_cnt==
0"....v=0..o=username 0 0 IN IP4 200.69.228.49..s=session..c=IN IP4
200.69.228.49..t=0 0..m=audio 58516 RTP/AVP 0..a=rtpmap:0 P
CMU/8000..
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
ACK sip:200@200.69.228.49:5060 SIP/2.0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;user=phone>;tag=3788
800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq: 1 ACK..User-Agen
t: Cisco ATA v2.15 ata18x (020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
ACK sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag=37
88800848;lr=on>..Via: SIP/2.0/UDP 200.69.228.49;branch=0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;us
er=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq
: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x (020927a)..Content-Length:
0....
#
U 200.69.228.49:5060 -> 200.69.228.52:5060
SIP/2.0 200 OK..Record-Route:
<sip:200@200.69.228.49;ftag=3788800848;lr=on>..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:1
00@200.69.228.49;user=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@
200.69.228.52..CSeq: 1 INVITE..Contact:
<sip:200@200.69.228.49:5090>..Content-Type: application/sdp..Server: Sip
EXpress router
(0.8.12 (i386/freebsd))..Content-Length: 131..Warning: 392
200.69.228.49:5090 "Noisy feedback tells: pid=956 req_src_ip=200.6
9.228.49 req_src_port=5060 in_uri=sip:200@200.69.228.49:5090;user=phone
out_uri=sip:200@200.69.228.49:5090;user=phone via_cnt==
0"....v=0..o=username 0 0 IN IP4 200.69.228.49..s=session..c=IN IP4
200.69.228.49..t=0 0..m=audio 58516 RTP/AVP 0..a=rtpmap:0 P
CMU/8000..
#
U 200.69.228.52:5060 -> 200.69.228.49:5060
ACK sip:200@200.69.228.49:5060 SIP/2.0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;user=phone>;tag=3788
800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq: 1 ACK..User-Agen
t: Cisco ATA v2.15 ata18x (020927a)..Content-Length: 0....
#
U 200.69.228.49:5060 -> 200.69.228.39:5060
ACK sip:200@200.69.228.39:5060;user=phone;transport=udp
SIP/2.0..Max-Forwards: 10..Record-Route: <sip:200@200.69.228.49;ftag=37
88800848;lr=on>..Via: SIP/2.0/UDP 200.69.228.49;branch=0..Via: SIP/2.0/UDP
200.69.228.52:5060..From: <sip:100@200.69.228.49;us
er=phone>;tag=3788800848..To:
<sip:200@200.69.228.49;user=phone>;tag=000003C941C67EA6..Call-ID:
3273316177@200.69.228.52..CSeq
: 1 ACK..User-Agent: Cisco ATA v2.15 ata18x (020927a)..Content-Length:
0....
^Cexit
10857 received, 0 dropped
Thank you
Juan
----- Original Message -----
From: "Jiri Kuthan" <jiri(a)iptel.org>
To: "Juan" <juan(a)korreo.com.ar>ar>; <serusers(a)lists.iptel.org>
Sent: Monday, February 23, 2004 7:06 PM
Subject: Re: [Serusers] voice mail doesn't stop redording and never sends
voicemail - need help
Have you seen the BYE on the net? And a reply to it?
-jiri
At 10:38 PM 2/23/2004, Juan wrote:
>Hi all,
>I have been trying to solve this problem for a couple of hours, but i
could'n. Could somebody help me?
>I have configure a voice mail system
(<http://iptel.org/~faqomatic/fom-serve/cache/92.html>http://iptel.org/~faqo
matic/fom-serve/cache/92.html) and I have a problem when it answer the call;
it doesn't stop recording when I hang on the phone and never sends the mail.
this is the sems debug:
---------------------------------------------------------------------------
---------------------------------------------------------------------
>(815) DEBUG: onSessionStart
(AnswerMachine.cpp:170): starting recording
message...(/tmp/2870083197(a)200.69.228.52.wav)
>(815) DEBUG: onSessionStart
(AnswerMachine.cpp:173): ... finished
recording message
>(815) DEBUG: sendToFIFO (AmRequest.cpp:199):
msg=<:t_uac_dlg:0000032F15FB7483
> >BYE
> >sip:100@200.69.228.52:5060;user=phone;transport=udp
> >sip:200@200.69.228.49;ftag=450631392;lr=on
> >From: <sip:200@200.69.228.49;user=phone>;tag=0000032F41C67EA6
> >To: <sip:100@200.69.228.49;user=phone>;tag=450631392
> >CSeq: 2 BYE
> >Call-ID: <mailto:2870083197@200.69.228.52>2870083197@200.69.228.52
> >Contact: <sip:200@200.69.228.49:5090>
> >Route: <sip:200@200.69.228.49;ftag=450631392;lr=on>
> >.
> >.
> >
> >>
> >(815) DEBUG: write_to_fifo (AmRequest.cpp:708): Write to fifo: completed
---------------------------------------------------------------------------
------------------------
>
>
>it never never gets the:
>
>DEBUG: getReturnCode (AmRequest.cpp:421): response from Ser: 200
fifo_t_reply
succeeded
>
>and sends the mail. When I call the voice mail with an ATA pointing to
port
5090 (SER voice mail) it works ok and sends the mail.
>The system is configured with 2 SER, one for
routing and the other for
voice mail.
Find attached
the configuration files.
Thank you in advance
Juan Priotti
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
5:12 p.m.
New subject: [Serusers] voice mail doesn't stop redording and never sendsvoicemail - need help
Hi Juan,
could you send us again your config files (the files in the current mail
empty...).
Further, it would be interesting to look at Ser's log file. If there are any
routing problems, Ser's timer should hit and send an error back to Sems.
Sems has also a timer, but much too long for that (about 5 minutes...).
-Raphael.
----- Original Message -----
From: "Juan" <juan(a)korreo.com.ar
To: <serusers(a)lists.iptel.org
Sent: Monday, February 23, 2004 10:38 PM
Subject: [Serusers] voice mail doesn't stop redording and never
sendsvoicemail - need help
Hi all,
I have been trying to solve this problem for a couple of hours, but i
could'n. Could somebody help me?
I have configure a voice mail system
(http://iptel.org/~faqomatic/fom-serve/cache/92.html) and I have a problem
when it answer the call; it doesn't stop recording when I hang on the phone
and never sends the mail. this is the sems debug:
----------------------------------------------------------------------------
--------------------------------------------------------------------
(815) DEBUG: onSessionStart (AnswerMachine.cpp:170): starting recording
message...(/tmp/2870083197(a)200.69.228.52.wav)
(815) DEBUG: onSessionStart (AnswerMachine.cpp:173): ... finished recording
message
(815) DEBUG: sendToFIFO (AmRequest.cpp:199):
msg=<:t_uac_dlg:0000032F15FB7483
BYE
sip:100@200.69.228.52:5060;user=phone;transport=udp
sip:200@200.69.228.49;ftag=450631392;lr=on
From: <sip:200@200.69.228.49;user=phone>;tag=0000032F41C67EA6
To: <sip:100@200.69.228.49;user=phone>;tag=450631392
CSeq: 2 BYE
Call-ID: 2870083197(a)200.69.228.52
Contact: <sip:200@200.69.228.49:5090
Route: <sip:200@200.69.228.49;ftag=450631392;lr=on
.
.
(815) DEBUG: write_to_fifo (AmRequest.cpp:708):
Write to fifo: completed
----------------------------------------------------------------------------
-----------------------
it never never gets the:
DEBUG: getReturnCode (AmRequest.cpp:421): response from Ser: 200
fifo_t_reply succeeded
and sends the mail. When I call the voice mail with an ATA pointing to port
5090 (SER voice mail) it works ok and sends the mail.
The system is configured with 2 SER, one for routing and the other for voice
mail.
Find attached the configuration files.
Thank you in advance
Juan Priotti
----------------------------------------------------------------------------
----
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
6:16 p.m.
New subject: [Serusers] voice mail doesn't stop redording and never sendsvoicemail - need help
Hi Raphael,
I'm attaching the config files and sems debug log. I'm sending you the ser's
log file in some minutes.
Thank you
Juan
----- Original Message -----
From: "Raphael Coeffic" <rco(a)iptel.org>
To: "Juan" <juan(a)korreo.com.ar>ar>; <serusers(a)lists.iptel.org>
Sent: Monday, February 23, 2004 7:12 PM
Subject: Re: [Serusers] voice mail doesn't stop redording and never
sendsvoicemail - need help
Hi Juan,
could you send us again your config files (the files in the current mail
empty...).
Further, it would be interesting to look at Ser's log file. If there are
any
routing problems, Ser's timer should hit and send
an error back to Sems.
Sems has also a timer, but much too long for that (about 5 minutes...).
-Raphael.
----- Original Message -----
From: "Juan" <juan(a)korreo.com.ar>
To: <serusers(a)lists.iptel.org>
Sent: Monday, February 23, 2004 10:38 PM
Subject: [Serusers] voice mail doesn't stop redording and never
sendsvoicemail - need help
Hi all,
I have been trying to solve this problem for a couple of hours, but i
could'n. Could somebody help me?
I have configure a voice mail system
(http://iptel.org/~faqomatic/fom-serve/cache/92.html) and I have a problem
when it answer the call; it doesn't stop recording when I hang on the
phone
and never sends the mail. this is the sems debug:
--------------------------------------------------------------------------
--
--------------------------------------------------------------------
(815) DEBUG: onSessionStart (AnswerMachine.cpp:170): starting recording
message...(/tmp/2870083197(a)200.69.228.52.wav)
(815) DEBUG: onSessionStart (AnswerMachine.cpp:173): ... finished
recording
message
(815) DEBUG: sendToFIFO (AmRequest.cpp:199):
msg=<:t_uac_dlg:0000032F15FB7483
BYE
sip:100@200.69.228.52:5060;user=phone;transport=udp
sip:200@200.69.228.49;ftag=450631392;lr=on
From: <sip:200@200.69.228.49;user=phone>;tag=0000032F41C67EA6
To: <sip:100@200.69.228.49;user=phone>;tag=450631392
CSeq: 2 BYE
Call-ID: 2870083197(a)200.69.228.52
Contact: <sip:200@200.69.228.49:5090>
Route: <sip:200@200.69.228.49;ftag=450631392;lr=on>
.
.
(815) DEBUG: write_to_fifo
(AmRequest.cpp:708): Write to fifo: completed
--------------------------------------------------------------------------
--
-----------------------
it never never gets the:
DEBUG: getReturnCode (AmRequest.cpp:421): response from Ser: 200
fifo_t_reply succeeded
and sends the mail. When I call the voice mail with an ATA pointing to
port
5090 (SER voice mail) it works ok and sends the mail.
The system is configured with 2 SER, one for routing and the other for
voice
mail.
Find attached the configuration files.
Thank you in advance
Juan Priotti
--------------------------------------------------------------------------
--
----
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
7:57 p.m.
New subject: [Serusers] voice mail doesn't stop redording and never sendsvoicemail - need help
Raphael,
Please find attached the ser log file.
Thanks
Juan
----- Original Message -----
From: "Raphael Coeffic" <rco(a)iptel.org>
To: "Juan" <juan(a)korreo.com.ar>ar>; <serusers(a)lists.iptel.org>
Sent: Monday, February 23, 2004 7:12 PM
Subject: Re: [Serusers] voice mail doesn't stop redording and never
sendsvoicemail - need help
Hi Juan,
could you send us again your config files (the files in the current mail
empty...).
Further, it would be interesting to look at Ser's log file. If there are
any
routing problems, Ser's timer should hit and send
an error back to Sems.
Sems has also a timer, but much too long for that (about 5 minutes...).
-Raphael.
----- Original Message -----
From: "Juan" <juan(a)korreo.com.ar>
To: <serusers(a)lists.iptel.org>
Sent: Monday, February 23, 2004 10:38 PM
Subject: [Serusers] voice mail doesn't stop redording and never
sendsvoicemail - need help
Hi all,
I have been trying to solve this problem for a couple of hours, but i
could'n. Could somebody help me?
I have configure a voice mail system
(http://iptel.org/~faqomatic/fom-serve/cache/92.html) and I have a problem
when it answer the call; it doesn't stop recording when I hang on the
phone
and never sends the mail. this is the sems debug:
--------------------------------------------------------------------------
--
--------------------------------------------------------------------
(815) DEBUG: onSessionStart (AnswerMachine.cpp:170): starting recording
message...(/tmp/2870083197(a)200.69.228.52.wav)
(815) DEBUG: onSessionStart (AnswerMachine.cpp:173): ... finished
recording
message
(815) DEBUG: sendToFIFO (AmRequest.cpp:199):
msg=<:t_uac_dlg:0000032F15FB7483
BYE
sip:100@200.69.228.52:5060;user=phone;transport=udp
sip:200@200.69.228.49;ftag=450631392;lr=on
From: <sip:200@200.69.228.49;user=phone>;tag=0000032F41C67EA6
To: <sip:100@200.69.228.49;user=phone>;tag=450631392
CSeq: 2 BYE
Call-ID: 2870083197(a)200.69.228.52
Contact: <sip:200@200.69.228.49:5090>
Route: <sip:200@200.69.228.49;ftag=450631392;lr=on>
.
.
(815) DEBUG: write_to_fifo
(AmRequest.cpp:708): Write to fifo: completed
--------------------------------------------------------------------------
--
-----------------------
it never never gets the:
DEBUG: getReturnCode (AmRequest.cpp:421): response from Ser: 200
fifo_t_reply succeeded
and sends the mail. When I call the voice mail with an ATA pointing to
port
5090 (SER voice mail) it works ok and sends the mail.
The system is configured with 2 SER, one for routing and the other for
voice
mail.
Find attached the configuration files.
Thank you in advance
Juan Priotti
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5 comments
3 participants
participants (3)
-
Jiri Kuthan -
Juan -
Raphael Coeffic