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I'm sorry, i send mail in asterisk-bug, but asterisk-users is better
for my problem<br>
<br>
Hello,<br>
<br>
During a call with canreinvite = no, at the beginning of the call I
lose 2 seconds of audio. <br>
is obvious when I call autoattendant.<br>
<br>
schema:<br>
SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk
1.4.24.1) --> Operator SIP<br>
<br>
capture of voip1:<br>
<br>
- Executing [0825387205@incoming_clients:1]
Dial("SIP/toto.fr-28fdf000", "SIP/0825387205@sipoperator") in new stack<br>
-- Called 0825387205@sipoperator<br>
-- SIP/sipoperator-28fed000 is making progress passing it to
SIP/toto.fr-28fdf000<br>
-- SIP/sipoperator-28fed000 is ringing<br>
-- SIP/sipoperator-28fed000 answered SIP/toto.fr-28fdf000<br>
-- Packet2Packet bridging SIP/toto.fr-28fdf000 and
SIP/sipoperator-28fed000 (((*****AUDIO IS CUT DURING 2 TO 3
SECONDS*****)))<br>
== Spawn extension (incoming_clients, 0825387205, 1) exited non-zero
on 'SIP/toto.fr-28fdf000'<br>
<br>
Native Bridging it's same problem.<br>
<br>
it's sip module bug ?? <br>
<br>
<div id="result_box" dir="ltr">When capturing with wireshark, at the
beginning of sound file, we see a break in sound.</div>
<a class="moz-txt-link-abbreviated" href="mailto:m.hubert@hexanet.fr"></a>
<br>
<div style="text-align: left;" id="result_box" dir="ltr">thank you in
advance</div>
<br>
<br>
sip conf:<br>
<br>
[general]<br>
port=5060<br>
bindaddr=0.0.0.0<br>
srvlookup=yes<br>
canreinvite=no<br>
rtcachefriends=yes<br>
directrtpsetup=no<br>
maxexpiry=300<br>
bridge=yes<br>
defaultexpiry=300<br>
useragent=toto<br>
<br>
PJ: shema of call with wireshark<br>
<br>
<br>
<br>
<br>
<br>
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