[asterisk-users] Problem loss 2 seconds audio when Packet2Packet bridging

[Hubert Mickael]

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 [0825387205 at incoming_clients:1] Dial("SIP/toto.fr-28fdf000", 
"SIP/0825387205 at sipoperator") in new stack
    -- Called 0825387205 at 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





-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090626/a978406c/attachment.htm 
-------------- next part --------------
An embedded and charset-unspecified text was scrubbed...
Name: graph_tel.txt
Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20090626/a978406c/attachment.txt