[asterisk-users] G.729 codec between avaya and asterisk

[Anselm Martin Hoffmeister]

Am Dienstag, den 23.10.2007, 02:56 -0700 schrieb satish patel:
> Dear all
>  
>         i have asterisk connected with avaya through E1 back-2-back
> now when i configure my sip client with g.729 codec then i m not able
> to put call from asterisk to avaya and when i user g.711 it is working
> fine so i dont know why i need G.729 on E1 Trunk it is TDM
> technologies then why my call fail in g.729 case 

Hi Satish,

Neither do I know why you _need_ G.729. Are there any specific reasons
why you do not want to use G711 in the sip client, which is "working
fine"? (Nota bene: there are some more codecs supported by asterisk,
some of which may be also supported by your sip phone)

Your E1 trunk obviously is G711-only - this is to be expected, because
the G711 wave samples are those which go over the wire (as time-division
multiplexed bitstream).

Together with the information from
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing

namely
** G.729 requires a license per channel unless it is used
** in pass-thru mode.

which exactly matches your setup (by the way that was the first google
match for "g729 asterisk") we can guess that you did not buy the license
which would be necessary for asterisk to transcode G729/G711.

> [sip_phone]------[asterisk]-----E1----[Avaya]----[analog_phone]
>  
> Asterisk sip client configure with g.711 alaw/ulaw
> Avaya phone client configure g.711 alaw/ulaw
>  
> suggest how do it implement g.729 on this case what change i have to
> done on both part

Avaya / E1 stays as is, sip client stays as is, your credit card data is
transferred to digium, and their license goes into the appropriate file
on your asterisk machine hard drive.

Others may have real world experience with those steps, but that is what
I read on this mailing list.

YMMV,
Anselm