[asterisk-users] Asterisk cuts off intial voice path on bridging SIP channel

[Mayur]

I am using asterisk-1.4.21 and it is configured to pass media through it for
SIP calls. I have observed that if the callee answers the call and starts
speaking immediately for e.g. 'Hello one two three', the caller would get to
hear only 'one two three'. From packet captures I can see that asterisk
receives all the RTPs from the callee but it truncates the 'Hello' word from
the voice path when passing the stream on the other side.

The signaling gets complete between caller and callee, so asterisk should
bridge the channels immediately. I am using canreinvite=no and nat=yes
option in sip.conf.

Has anyone observed this issue why asterisk is cutting of the initial voice?


 

---Mayur

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