[Asterisk-Users] Asterisk terminates unexpectedly with SIP call and G.723 codec

[Paul Cheng]

Hi,

I'm using a Cisco ATA186 and iConnect to complete PSTN calls to the US. 
I've noticed that when I set the Cisco ATA to use LBRCodec to 0 (g.723 
instead of g.729), AudioMode 0x00150015 and RxCodec, LxCodec to 0, (use 
g.723) Asterisk will connect to iConnect, successfully natively bridge 
the call and then about two seconds later not just drop the call, but 
terminate unexpectedly.

The asterisk daemon will stop uncleanly.

If I set the RxCodec, LxCodec back to 2 (g.711ulaw), then it works 
fine. In my sip.conf, I have allow=all.

Has anyone else run into this problem?