[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:
[quote][callcentric]
; 9/9/11: attempt to restrict codec
;disallow=all
;disallow=ulaw
;disallow=alaw
;
; G729
: ====
allow=g729 ;enables G729 outbound: commenting results in busy signal >>asterisk -rx “sip show channels”
; commenting line above yields: WARNING[7854]: chan_sip.c:5368 sip_call: No audio format found to offer. Cancelling call to 1777XXXXXXXX
;
type=peer
context=from-callcentric ;how incoming calls are handle: defined in: extensions.conf host=callcentric.com
defaultuser=1777XXXYYY ;account=astrix, pw=laptop+1
secret=MyPassword
fromuser=1777XXXYYY fromdomain=callcentric.com
insecure=invite ;9/6/11[/quote]
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.
Thanks. I did not know this: where did you read this? Are you sure this is right? I ask because I forced both ATA lines to use g729 and they connected to one another. ‘sip show channels’ revealed that the call used g729 at both ends:
athomehost*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
192.168.8.110 101 326829a1-8e7e77 0x100 (g729) No Tx: ACK 101
204.11.192.34 1777XXXYYYY 0f0e3a7f7028818 0x100 (g729) No Tx: ACK callcentri
TESTING WITH ONE INSTANCE OF G729
I decided to test the notion that only one instance of G729 can be in operation: I tried shutting off the ATA’s Line 1 and induce an inbound call to L2 (Google Voice → PSTN → via callcentric → Asterisk → L2).
I shutdown Line 1 and had Google Voice call Line 2 and connect it to a 1-800 number. This call went through but was using ulaw codec (I was expecting it to use G729a, the preferred codec). I went back into the ATA and forced line 2 to use the G729A codec and repeated the GV 1-800 call. This time, the inbound through asterisk failed as described in the Original Post.
Actually there is some contradictory information on this, some sites state it supports only 1 G.729 channel and others state it supports two. Initially I got it confused with the PAP2T. I’m not sure how many channels does the SPA2102 support, so please excuse me if I mislead you.
Could you please post your dialplan here? I think it could help tracing the issue. Also a SIP debug would help a lot.