[asterisk-users] Zap Codec

Tue Apr 15 13:56:24 CDT 2008

Correct, those are two peers talking direct, one call leg (SIP->SIP).

In this case, you have two call legs which are then bridged:
SIP -> Asterisk
Asterisk -> Zap

You've already negotiated g729 before Asterisk notices that the call is
going out Zap (via your dialplan). At this point, you have to transcode
if your peer is set to use g729. Otherwise, force your SIP end to talk
ulaw.

-----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 11:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Codec

Correct, but if I have two sip peers, one with G729&ulaw, the other with
gsm&ulaw, they will negotiate before trying to send audio.

With ZAP, it tries to transcode whatever it receives into ulaw, period.
No negotiation to even tell the client to send ulaw if capable.

With no call level control(or dialplan logic, or anything!), I either
use ulaw for ALL CALLS from sip peers(to other sip peers, to iax peers,
to ZAP peers/channels), or use a combination of codecs and make sure
it's able to be transcoded for the ZAP channels.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Darryl
Dunkin
Sent: Tuesday, April 15, 2008 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zap Codec

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:
> 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:
>> On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote:
>>> But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I
only want
>>> ulaw used when SIPPEER-ZAP is the case.
>> Set(_SIP_CODEC=ulaw)
>> Dial(Zap/g0/...)
>>
>
> --
> Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN,
QoS,
> T-1, PRI, Frame Relay, Linux, and network design.  Based near
> Birmingham, AL.  Now accepting clients worldwide.
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--
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T-1, PRI, Frame Relay, Linux, and network design.  Based near
Birmingham, AL.  Now accepting clients worldwide.

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