[asterisk-users] One Way Audio After Dial

[Norman Franke]

I've encountered an odd situation with Asterisk 1.4.19 that I can't  
figure out.

If I dial an extension via a Cisco AS5400 with the "g" option to come  
back, when I then Dial another extension after that, we don't get  
audio from the caller. There are no firewalls, no routers, no  
anything but a network switch between. The calls come in as SIP from  
the Cisco and terminate on a SIP soft client.

I searched for something similar, but everything I found dealt with  
NATing and the like, which I don't do. Static IPs to static IPs.

If I remove the first dial with the "g", then everything works just  
fine. If I call a local SIP soft client, everything works fine  
(instead something via the Cisco.)

If I set "canreinvite=no" for the Cisco everything works. It seems  
like the "g" option should disable canreinvite for that call, so why  
the difference?

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

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