[Asterisk-Users] iconnecthere 480 error: is there a workaround?

[Gregg Lebovitz]

Luke,

here's some information I got back from iconnect:

1) the 7777 prefix is not a toggle. It tells iconnects SIP gateway to
use compressed codecs. The choices are gsm, g723.1, g729.

If you don't use 7777, the gateway will tried to use PCMu/8000 (ulaw?)
or PCMa/8000 (alaw?).

I can get the gateway to work with g723.1 and gsm, but I can't get it to
work with ulaw or alaw. My phone device is a quicknet linecard.

The g723.1 format on the linecard does not work with iconnect. If I use
it then the audio to and from iconnect is distorted (as if it is using
the wrong format or has sampling errors).

Gregg

On Thu, 2003-03-20 at 18:25, Luke Howard wrote:
> I've found the same.
> 
> If I make an outgoing call (snom 200 handset), I get about 5 seconds
> of audio and then it drops out (very occasionally it does work).
> 
> Incoming calls appear to work, though.
> 
>   -- Executing Goto("SIP/515-Office-143b", "iconnecthere-ulaw|91800XXXXXXX|1") in new stack
>   -- Goto (iconnecthere-ulaw,91800XXXXXXX,1)
>   -- Executing StripMSD("SIP/515-Office-143b", "1") in new stack
>   -- Executing Dial("SIP/515-Office-143b", "SIP/1800XXXXXXX at iconnecthere") in new stack
>   -- Called 1800XXXXXXX at iconnecthere
>   -- SIP/iconnecthere-960b answered SIP/515-Office-143b
>   -- Attempting native bridge of SIP/515-Office-143b and SIP/iconnecthere-960b
>   -- Got SIP response 480 "Temporarily not available" back from 213.137.73.178
>  == Spawn extension (iconnecthere-ulaw, 1800XXXXXXX, 2) exited non-zero on 'SIP/515-Office-143b'
> 
> SIP config is:
> 
> [general]
> port=5060
> bindaddr=0.0.0.0
> context=sip-remote
> disallow=all
> allow=ulaw
> allow=alaw
> tos=lowdelay
> tos=184
> register => 1XXXXXXXXXX:XXXX at natrelay.deltathree.com
> 
> [iconnecthere]
> type=friend
> username=XXXXXXXX
> password=XXXX
> host=sipauth.deltathree.com
> context=iconnecthere-ulaw
> callerid="PADL Software Pty Ltd" <(XXX) XXX XXXX>
> ;txgain = 5.0;
> ;rxgain = 5.0;
> inbanddtmf=1
> 
> -- Luke
> 
> P.S. Is anyone planning on licensing G 723.1 for use with Asterisk? As
> I understand it, buying a LineJACK won't suffice if the card's DSP is
> not actually used.
> --
> Luke Howard | PADL Software Pty Ltd | www.padl.com
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