Displaying 1 result from an estimated 1 matches for "incoming_clients".
2009 Jun 26
0
Problem loss 2 seconds audio when Packet2Packet bridging
...blem
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-28fed00...