I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for
my problem
Hello,
During a call with canreinvite = no, at the beginning of the call I lose
2 seconds of audio.
is obvious when I call autoattendant.
schema:
SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1.4.24.1)
--> Operator SIP
capture of voip1:
- Executing [0825387...@incoming_clients:1] Dial("SIP/toto.fr-28fdf000",
"SIP/0825387...@sipoperator") in new stack
-- Called 0825387...@sipoperator
-- SIP/sipoperator-28fed000 is making progress passing it to
SIP/toto.fr-28fdf000
-- SIP/sipoperator-28fed000 is ringing
-- SIP/sipoperator-28fed000 answered SIP/toto.fr-28fdf000
-- Packet2Packet bridging SIP/toto.fr-28fdf000 and
SIP/sipoperator-28fed000 (((*****AUDIO IS CUT DURING 2 TO 3 SECONDS*****)))
== Spawn extension (incoming_clients, 0825387205, 1) exited non-zero
on 'SIP/toto.fr-28fdf000'
Native Bridging it's same problem.
it's sip module bug ??
When capturing with wireshark, at the beginning of sound file, we see a
break in sound.
thank you in advance
sip conf:
[general]
port=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no
rtcachefriends=yes
directrtpsetup=no
maxexpiry=300
bridge=yes
defaultexpiry=300
useragent=toto
PJ: shema of call with wireshark
|Time | 81.XX.XX.XX | 83.XX.XX.XX |
|2,820 | INVITE SDP ( telephone-event) |SIP From:
sip:[email protected] To:sip:[email protected]
| |(5060) ------------------> (5060) |
|2,827 | 401 Unauthorized |SIP Status
| |(5060) <------------------ (5060) |
|2,827 | ACK | |SIP Request
| |(5060) ------------------> (5060) |
|2,827 | INVITE SDP ( telephone-event) |SIP From:
sip:[email protected] To:sip:[email protected]
| |(5060) ------------------> (5060) |
|2,835 | 100 Trying| |SIP Status
| |(5060) <------------------ (5060) |
|3,326 | 183 Session Progress SDP ( telephone-event) |SIP
Status
| |(5060) <------------------ (5060) |
|3,390 | RTP (g711U) |RTP Num packets:57
Duration:1.119s SSRC:0x1F8C28DA
| |(18344) ------------------> (12432) |
|4,243 | RTP (g711U) |RTP Num packets:15
Duration:0.280s SSRC:0x2CA92AC1
| |(18344) <------------------ (12432) |
|4,322 | 180 Ringing |SIP Status
| |(5060) <------------------ (5060) |
|4,525 | 200 OK SDP ( telephone-event) |SIP Status
| |(5060) <------------------ (5060) |
|4,526 | ACK | |SIP Request
| |(5060) ------------------> (5060) |
|4,530 | RTP (g711U) |RTP Num packets:9
Duration:0.159s SSRC:0x1B7B
| |(18344) ------------------> (12432) |
|4,883 | RTP (g711U) |RTP Num packets:107
Duration:2.119s SSRC:0x1D3848FA
| |(18344) ------------------> (12432) |
|5,041 | RTP (g711U) |RTP Num packets:125
Duration:1.980s SSRC:0x3C75BBDB
| |(18344) <------------------ (12432) |
|7,026 | BYE | |SIP Request
| |(5060) ------------------> (5060) |
|7,032 | 200 OK | |SIP Status
| |(5060) <------------------ (5060) |
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