[Asterisk-biz] Codec error connecting to cisco gateway

[Brian Fertig]

It will work.  Those changes need to be made to the cisco.  Yes g729r8 will
work with a cisco.  I do it now.  

You need to get with your client and see what is configuration for the
dialpeer is.  Paste it here minus IP's and such so I can see what it looks
like.  And if I could ask..  If it's a cisco why not just use SIP instead of
H323?   All it is, is a matter of putting

session protocol sipv2

In your dialpeer.

But your previous question to the G729 issue is simple to resolve.   You
just need to make sure they match.  Try sticking this in the cisco's
dialpeer configuration.

codec g729r8 bytes 40

This should fix the issue.

b.

-----Original Message-----
From: asterisk-biz-bounces at lists.digium.com
[mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Zafer Khodr
Sent: Sunday, November 13, 2005 12:14 AM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: RE: [Asterisk-biz] Codec error connecting to cisco gateway

Thanks for the Prompt reply.
I got a little mixed up with your solution...
First of all We are not on the same network so we cant use g711.
I wouldn't even know what a cisco looks like, the cisco is what my potential
customer is using.

Quoting from your reply "

For example:

codec g729r8 

isn't right.  You should use something like this: 

codec g729r8 bytes 40  "

So are you saying that G729r8 wont work??
If not do the modifications need to be made on * or on the cisco?
What is the best way to make g729 work.

Thanks
Zafer

-----Original Message-----
From: asterisk-biz-bounces at lists.digium.com
[mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Brian Fertig
Sent: Sunday, 13 November 2005 4:02 PM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: RE: [Asterisk-biz] Codec error connecting to cisco gateway

Being that I use a lot of cisco with my * box I would say it's a size error.
I got this awhile back and found you had to tell the cisco what size packet
to use.  

For example:

codec g729r8 

isn't right.  You should use something like this: 

codec g729r8 bytes 40

I have found this works very well with Digium's G729A codec.  You can also
setup a class for the codecs so that more than one codec can be used on a
dialpeer.  If your cisco and * are on the same lan I would recommend using
g711 it's the best quality you can get.


Brian


-----Original Message-----
From: asterisk-biz-bounces at lists.digium.com
[mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Zafer Khodr
Sent: Saturday, November 12, 2005 11:42 PM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: [Asterisk-biz] Codec error connecting to cisco gateway

I hope someone can help me with this.

I have a cisco gateway trying to dial my box via H.323.
The call comes through ok and gets routed properly.. only thing is " NO
AUDIO "
I am confident that I have narrowed down the problem to a codec issue.
I have the relevant G729 licences which were purchased from diguim.
The calling party was calling using G729.
I asked them which version of the codec and they told me G729r8 .  what ever
that means?????

I get about 150 of these lines in the log file per second while the call is
active.

Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.
Nov 12 00:47:53 DEBUG[24561]: OH323/R7907: Dropped zeroed G.729 frame.

Another party with a quantum device tried to call me and it was the same
error.. they even tried a SIP call and the result was the same.

I had one party call me through a softswitch and both sip and H323 worked
fine for that person.

In my mind it seem that there is some kind of incompatibility with the audio
codec the quintum and cisco are sending and the one asterisk is using.
Could this be the case and if so is there a work around?
Has anyone else had this issue before or does anyone know possibly if I am
wrong what the problem may be?

Thanks in advance


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