[Asterisk-Users] g729 codex experimentation

[Kim C. Callis]

Yesterday, I started to experiment with Cisco to Cisco SIP calls using
the g729 codec. According to the documentation, both the ATA-186 and
7960 are able to make use of the g729.

>From an earlier e-mail, I made a change to the configuration of the ATA,
changing the values:

LBRCodec:3
RxCodec: 3
TxCodec: 3

The first thing I noticed was that when I did a sip show channels, the
format had changed from ULAW to ALAW. The problem that I am having is
that when I attempt to call any of the 7960, I get the following:

    -- Executing Dial("SIP/1200-3bc9", "SIP/1000|50|rtm") in new stack
    -- Called 1000
    -- Called 1200
    -- Started music on hold, class 'default', on SIP/1200-3bc9
    -- Got SIP response 488 "Not Acceptable Here" back from 66.12.5.14
    -- Got SIP response 488 "Not Acceptable Here" back from 67.15.17.22

So is there something I need to set on the 7960s in order to make use of
that particular codec? Am I missing something in the compiling of *?

Any help would be very appreciated!

Kim Callis