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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:

[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.

This article indicates that two G729 Phones can use Asterisk as passthrough and without asterisk having to transcode.

The SPA2102 does support g729, but it has only 1 license, so only 1 g729 call at a time. You can’t have 2 simultaneous calls with this ATA using g729.

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.

Sip.conf:

[quote][callcentric]
; 9/9/11: attempt to restrict codec
disallow=all
allow=g729
;allow=ulaw
;allow=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 1777XXXXXXX
;
type=peer
context=from-callcentric ;how incoming calls are handle: defined in: extensions.conf
host=callcentric.com
defaultuser=1777XXXYYYY ;account=astrix, pw=laptop+1
secret=myPassword
fromuser=1777XXXYYYY
fromdomain=callcentric.com
insecure=invite ;9/6/11

[101]
allow=g729; necessary to call other callcentric line:
allow=ulaw
type=friend
host=dynamic
nat=yes
qualify=yes
context=mario-default
defaultuser=101
secret=myPassword
callerid=“SPA2102 L2” <101>
mailbox=101[/quote]

Inbound callcentric excerpt from extension.conf:

[from-callcentric]
;exten => s, 1, Dial(SIP/101, 180, D(:1)); fail
exten => 1777XXXYYYY,1,Dial(SIP/101)

I comment extensivel about the ‘s’ extension not working here: http://forums.digium.com/viewtopic.php?f=1&t=79929