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[ddunkin at netos.net]

This is SIP channel you need to limit. Forcing ulaw only, the Polycom
will fall back to ulaw.

Per peer, in your sip.conf:
disallow=all
allow=ulaw

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jeremy
Mann
Sent: Monday, April 14, 2008 14:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Zap Codec

Is there a way to force Zap channels to only use ulaw, and not even
attempt g729 negotiation?



My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm
not licensed for the codec on the asterisk box.


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[eric at fnords.org]

The PSTN only allows ulaw or alaw (depending on your location). You
CANNOT send calls in any other codec over a PSTN line. Generally, if
you want to use G729 then you must buy a G729 license (with a few
exceptions).

Jeremy Mann wrote:

[jmann at txhmg.com]

So in other words, if I have G729 enabled on the phones, I must get G729 licenses to use Zap channels. Otherwise I have to use ULAW for everything?

I fail to understand why it'd be difficult to do codec negotiation on SIP-ZAP calls, Zap sends that it only supports ulaw, if the phone doesn't then the call is cancelled or forwarded to logic to translate.

I realize G729 is fairly cheap, but it's useless server overhead when the phone supports the codec it needs natively.

Is there any dialplan logic that can coerce the transaction to be ulaw only? Setting something in the SIP header perhaps?
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
Sent: Tuesday, April 15, 2008 8:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Codec

The PSTN only allows ulaw or alaw (depending on your location). You
CANNOT send calls in any other codec over a PSTN line. Generally, if
you want to use G729 then you must buy a G729 license (with a few
exceptions).

Jeremy Mann wrote:

[eric at fnords.org]

If you are talking between two g729 endpoints, the Asterisk overhead is
very small.

Jeremy Mann wrote:

[tilghman at mail.jeffa...]

On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote:

[eric at fnords.org]

That would work just spiffy if you are calling another SIP device, but
by the time the call gets to that point in the dialplan the codec of the
originating device has already been chosen and set in stone.

Tilghman Lesher wrote:

[jmann at txhmg.com]

Sadly you are correct:
-- Executing [8173104999 at from-sip:4] Set("SIP/156-083514c0", "_SIP_CODEC=ulaw") in new stack
-- Executing [8173104999 at from-sip:5] NoOp("SIP/156-083514c0", "4") in new stack
-- Executing [8173104999 at from-sip:6] NoOp("SIP/156-083514c0", """) in new stack
-- Executing [8173104999 at from-sip:7] Dial("SIP/156-083514c0", "") in new stack
[Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256
[Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [8173104999 at from-sip:8] Hangup("SIP/156-083514c0", "") in new stack

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
Sent: Tuesday, April 15, 2008 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Codec

That would work just spiffy if you are calling another SIP device, but
by the time the call gets to that point in the dialplan the codec of the
originating device has already been chosen and set in stone.

Tilghman Lesher wrote:

[eric at fnords.org]

If you want to get a G729 call to go via Zap you must purchase a G729
license. No amount of discussion is going to change that.

Jeremy Mann wrote:

[jmann at txhmg.com]

I guess that's my frustration, I don't want it g729, I want it ulaw, I just wish Zap did codec negotiation from the client. It'd be a nice option instead of automatically trying to translate if it's not ulaw. Could save some processor overhead(obviously at the expense of bandwidth).

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
Sent: Tuesday, April 15, 2008 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Codec

If you want to get a G729 call to go via Zap you must purchase a G729
license. No amount of discussion is going to change that.

Jeremy Mann wrote:

[ddunkin at netos.net]

Asterisk builds two channels and bridges them together. If the codecs
mis-match then it must transcode, the negotiation on the Zap end is done
seperately from the SIP end, so it does not care what your handset
decided on.

If you want ulaw, use ulaw, not g729 (on any call leg). You won't be
able to mix and match codecs between calls, choose one for all calls and
stick with it.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jeremy
Mann
Sent: Tuesday, April 15, 2008 08:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Codec

I guess that's my frustration, I don't want it g729, I want it ulaw, I
just wish Zap did codec negotiation from the client. It'd be a nice
option instead of automatically trying to translate if it's not ulaw.
Could save some processor overhead(obviously at the expense of
bandwidth).

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
Wieling
Sent: Tuesday, April 15, 2008 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Codec

If you want to get a G729 call to go via Zap you must purchase a G729
license. No amount of discussion is going to change that.

Jeremy Mann wrote:

[Guest]

In the sip peer definition,

disallow=all
allow=g729
allow=ulaw

SHOULD work. Asterisk can't transcode g729, so it should fall on ulaw
for the ZAP calls. But, when your polycoms talk with each other, as
long as all necessary REINVITEs happen, they should use the 729 codec I
believe. Remember however, that many options to the Dial application,
like t,w,m,k (or so) REQURE asterisk to remain in the media path.

moj

Jeremy Mann wrote:

[ex.vitorino at gmail.com]

On Tue, Apr 15, 2008 at 11:37 PM, Mojo with Horan & Company, LLC
<mojo at horanappraisals.com> wrote: