asterisk users - Oct 2004 - Working Wellgate *SIP* 38xx/35xx hardware anyone?

I'm loosing hair at cosmic speed now for the past 10 days.

Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to have 
very buggy firmware possibly due to hastely done porting from H.323 
firmware.

Is there anyone on this mailing list who was able to:

1. setup a 35xxA FXS with all ports authenticating properly with *?
or
2. setup a 38xx FXO to work as dial-in from pstn to * fixed sip number 
and dial-out to pstn with prefix change WHILE having normal two-way 
voice working? If I call from pstn to the FXO's #, I hear voice ok both 
ways, but if I dial out form a sip phone via FXO's line1 out to a phone, 
then sip phone's voice isn't heard on the pstn phone.

Also, FXO doesn't detect the busy signal properly, if I hangup pstn 
phone, i can hear the busy tone on the sip phone. And yes, I've done 
frequency analysis for the busy tone using Adobe Audition 1.5 
(ex-cooledit pro).

AND FINALLY, anyone has received ANY email from Welltech sales/support 
emails?

here is the snip from my extensions.conf, anything wrong?

exten => _3741XXXXXX,1,Dial(SIP/1111${EXTEN:4}@x.y.z.x,60)
exten => _3741XXXXXX,2,Congestion
exten => _3741XXXXXX,3,Hangup
exten => _3741XXXXXX,102,Congestion
exten => _3741XXXXXX,103,Hangup


regards,
Vahan

The 38xx FXO SIP firmware is buggy. Welltech is aware of this. They are 
working in a totally new SIP firmware version promised for a "near"
future.

Jorge Mendoza


Vahan Yerkanian wrote:
> I'm loosing hair at cosmic speed now for the past 10 days.
> 
> Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to
have
> very buggy firmware possibly due to hastely done porting from H.323 
> firmware.
> 
> Is there anyone on this mailing list who was able to:
> 
> 1. setup a 35xxA FXS with all ports authenticating properly with *?
> or
> 2. setup a 38xx FXO to work as dial-in from pstn to * fixed sip number 
> and dial-out to pstn with prefix change WHILE having normal two-way 
> voice working? If I call from pstn to the FXO's #, I hear voice ok both
> ways, but if I dial out form a sip phone via FXO's line1 out to a
phone,
> then sip phone's voice isn't heard on the pstn phone.
> 
> Also, FXO doesn't detect the busy signal properly, if I hangup pstn 
> phone, i can hear the busy tone on the sip phone. And yes, I've done 
> frequency analysis for the busy tone using Adobe Audition 1.5 
> (ex-cooledit pro).
> 
> AND FINALLY, anyone has received ANY email from Welltech sales/support 
> emails?
> 
> here is the snip from my extensions.conf, anything wrong?
> 
> exten => _3741XXXXXX,1,Dial(SIP/1111${EXTEN:4}@x.y.z.x,60)
> exten => _3741XXXXXX,2,Congestion
> exten => _3741XXXXXX,3,Hangup
> exten => _3741XXXXXX,102,Congestion
> exten => _3741XXXXXX,103,Hangup
> 
> 
> regards,
> Vahan
>

On 06/10/2004 17:29 Vahan Yerkanian said the following:
> 1. setup a 35xxA FXS with all ports authenticating properly with *?
> or
search this list's archives for a discussion on the wellgate 3504A SIP 
proxies. i've fashioned a patch to chan_sip.c which allows the 3504A to 
register all ports, but it works on freebsd only right now. you could try 
porting it to linux.

-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh@alphaque.com                (0 0)    http://www.alphaque.com/
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