Sir,
I have problem wirh Asterisk 1.4 when a sip user receive a call than its work fine but when sip user transfer call to another sip user than only one only one side audio only sip user hear the audio but souce will not hear anything
rajeev
Most one-way audio problems in SIP are firewall related. Start there (paying attention to the RTP ports).
Sir,
I have disable all firewall.
Rajeev.
Have you put the âtâ option at the end of the Dial command?
Yes sir i have use t option
Can all the SIP phones make and receive their own calls with no problems (directly dialled out - not transferred)?
Sir,
I have solve the problem when i have make canreinvite=no from canreinvite=yes than it works fine but it will crash Asterisk after some time
when i have make canreinvite=no.
Please help
Rajeev.
The need for canreinvite=no implies that you have network address translation.
However, as another thread indicates a very old version, you should really upgrade before investigating further.