[asterisk-users] Zap Codec

[Darryl Dunkin]

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
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--
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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intended recipient, further disclosures are prohibited without proper
authorization. If you are not the intended recipient, any disclosure,
copying, printing, or use of this information is strictly prohibited and
possibly a violation of federal or state law and regulations. If you
have received this information in error, please notify Texas Health
Management Group immediately at 1-817-310-4999. Texas Health Management
Group, its subsidiaries, and affiliates hereby claim all applicable
privileges related to this information.

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