[Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller

[John Baker]

Are you binding to one particular interface?

I.e., Is bindaddr set to something besides 0.0.0.0?

If so, ifdown your other interfaces, start asterisk, then bring up the other
interfaces.

If not, check your codecs.  You need some lines like this in sip.conf:

disallow=all                   ; Disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=alaw
allow=gsm

If that still doesn't do it, I'll bet you're working some sort of NAT.  Try
the latest release of asterisk in the CVS and check the archives of this
mail list for the new NAT parameters in sip.conf

John

----- Original Message ----- 
From: "Frankie Gravato" <nanog at cfsdigital.com>
To: "Asterisk" <asterisk-users at lists.digium.com>
Sent: Sunday, January 25, 2004 12:22 AM
Subject: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed
to caller


>
>
> I've  been  beating  my head for 5 hours to figure out why my asterisk
> server or sipura isn't passing my voice over to the caller. It seems i
> can  hear  the  caller  but  they  can't  hear  me it seems either the
> asterisk or the sipura isn't passing this information.
>
> Here's my setup specs
>
> asterisk  server  0.7.1  - X100P Card - Sipura 2000 - Nufone Service -
> Voicepulse Service and DID's
>
> when  i  get  Phone call using the Voicepulse or Pstn the caller can't
> hear  me  or  barely  hear me. The Sipura is running Firmware 1.20 and
> calls  are  being  passed  using  Ulaw  Codec? Anyone out there in the
> asterisk community please oh please help me before i do something that
> my asterisk server won't like.
>
>
>
>
>
>
> -- 
> Best regards,
> Frankie (fgravato at cfsdigital.com)
> mailto:nanog at cfsdigital.com
>
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