My system is setup as follows
[quote]SPA2102 ATA (Line2) <==> Asterisk 1.8.4 <==> Callcentric <==> SPA2102 ATA (Line1)[/quote]
Asterisk 1.8.4 and Line 1 are registered to separate Callcentric accounts. L2 is registered with the Asterisk server. Calls from L2 to Asterisk are successful and show that they are using the G729 codec. I attribute this to allowing g729 in both the Callcentric and Line2 entries in sip.conf:
; 9/9/11: attempt to restrict codec
allow=g729 ;enables G729 outbound: commenting results in busy signal >>asterisk -rx “sip show channels”
; commenting line above yields: WARNING: chan_sip.c:5368 sip_call: No audio format found to offer. Cancelling call to 1777XXXXXXXX
context=from-callcentric ;how incoming calls are handle: defined in: extensions.conf
defaultuser=1777XXXYYY ;account=astrix, pw=laptop+1
When calling from Line 2 to Asterisk, I am receiving a “user is unavailable message” and this error message:
L2 was using G729 only, so this means that inbound SIP calls to * could setup the call for G729. How / where do I configure Asterisk so that it will setup a g729 passthrough to my (G729 Compatible) ATA? It’s odd that outbound passthrough works, yet inbound passthrough fails.
Relaxing the G729 requirement on L1 results in a ulaw call between the two lines.