[asterisk-users] FW: G.726 on Asterisk 1.4.0

[Carlos Alperin]

 Ok,

With everything restore on rtp.c, still I have no audio however the call is
not destroyed immediately as before.

I'm going to put a second Granstream box, and findout if between two boxes
this happen too.

I cannot believe that we cannot do 2 g726 on the same box at one time.

Carlos

-----Original Message-----
From: Carlos Alperin [mailto:calperin at senecacom.net] 
Sent: Wednesday, December 06, 2006 11:16 AM
To: 'asterisk-users at lists.digium.com'
Subject: FW: G.726 on Asterisk 1.4.0
Importance: High

This is what I found today googling on the Web, also I post it now in order
to save time to others:


 The G726-32 codec:

* It has been determined that previous versions of Asterisk used the wrong
codeword packing order for G726-32 data. This version supports both
available packing orders, and can transcode between them. It also now
selects the proper order when negotiating with a SIP peer based on the codec
name supplied in the SDP. However, there are existing devices that
improperly request one order and then use another; Sipura and Grandstream
ATAs are known to do this, and there may be others. To be able to continue
to use these devices with this version of Asterisk and the G726-32 codec, a
configuration parameter called 'g726nonstandard' has been added to sip.conf,
so that Asterisk can use the packing order expected by the device (even
though it requested a different order). In addition, the internal format
number for G726-32 has been changed, and the old number is now assigned to
AAL2-G726-32. The result of this is that this version of Asterisk will be
able to interoperate over IAX2 with older versions of Asterisk, as long as
this version is told to allow 'g726aal2' instead of 'g726' as the codec for
the call.

Now, I can complete a call but I have now audio yet. I believe that I need
to restore the original rtp.c and recompile it.

I hope this help someone else...


Today before to find the new information:

Since nobody answer my previous question (It looks like g.726 is a bad
word).
 
I have this scenario:
 
One box with Asterisk 1.4.0 beta 2
 
IAX to anothers Asterisk working properly.
 
As an ATA I have only one Grandstream HT496.
 
Two lines on the ATA 727 & 726.
 
>From outside I can call any of those two extensions if:
 
I defined both as ulaw (G.711)
One as ulaw and the other as G.729
Only one at the time if I define both as G729 Only the G711 if I define one
as G726 and the other as G711.
 
No way to make calls between the two extensions if both are G726, or both
are G729 Or if one is G726 and the other is G729 or ulaw.
 
I allways get codec not match or invalid on the console.
 
Grandstream claims that they can handle two g729 calls at the time ( I never
was able to do this), and even when I change the rtp.c the G726 is not
working at all. (I 'm not sure but it looks like is more an Asterisk
problem, since the modification was made for Sipura boxes, but even when
other phones, after the 1.2.6 we have fast busy  problems on those devices
with that code)
 
So at this point. if someone has any other experience that want to share
I'll appretiate it.
 
Thanks
 
Carlos Alperin
 
-----Original Message-----
From: Carlos Alperin [mailto:calperin at senecacom.net]
Sent: Tuesday, December 05, 2006 2:29 PM
To: 'asterisk-users at lists.digium.com'
Subject: G.726 on Asterisk 1.4.0
Importance: High

 
I'm trying to make a new box with Asterisk 1.4.0, work with one ATA
GrandStream 496 and G.726.
 
However I modified the rtp.c as suggested for the Sipura's ATA with
USE_DEPRECATED_G726=1 is not working.
 
Someone knows about this?
 
Thanks,
 
Carlos Alperin