[Asterisk-Users] SIP routing over IAX2

[Bart J. Smit]

Thanks Alejandro,

I'm sure the codecs are fine, as I can make calls inbound to the LAN
Asterisk.

Can you tell me which configuration changes I need to make on each
Asterisk to route these calls?

Bart...

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alejandro
Vargas
Sent: 16 March 2006 08:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP routing over IAX2

2006/3/16, Bart J. Smit <bart at smits.co.uk>:
> Can Asterisk do this? I am relatively new to Asterisk. I guess I'm
after
> something like an email smarthost feature for SIP.

Yes, Asterisk can do protocol conversion as well as codec conversion.
Just configure phones and asterisk to connect correctly (i.e. echo
test working) and make sure the audio codecs you are using are
compatible or are enableded in asterisk.

I.E. One case that will not work: phone or trunk A: protocols
supported speex,iBLC. Asterisk: supports speex, iBLC, G711. Phone B:
supports G729, G723.

In this case, Asterisk should converted one of the codecs supported by
B to one of supported by A, but Asterisk can't decode them because you
don't installed any codec for G729 nor G723.

Cases it will work:
if A supports also G729 or G723: in this case, Asterisk don't need to
do transcoding, then it does not matter if it has tihs codecs.
If you install G729 and/or G723 in Asterisk. In this case, Asterisk
can decode the audio and re-encode with speex or iBLC.


--
Alejandro Vargas
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