asterisk users - Jan 2004 - 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@cfsdigital.com)                          
mailto:nanog@cfsdigital.com

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@cfsdigital.com>
To: "Asterisk" <asterisk-users@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@cfsdigital.com)
> mailto:nanog@cfsdigital.com
>
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>

Frankie Gravato wrote:
> 
> 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.
> 
> 
I just received my Sipura on Friday and have been testing it extensively 
over the weekend.  I have noticed an issue similar to what you mention 
above.  For the record, the sipura tells me I'm running software version 
1.0.20.  Also, there is NO nat configuration that is causing my problem.

When I receive a call over my X100P and dial my 3 SIP phones (one gs 
budgetone 100, two analong phones through sipura), if I answer the 
analong phone connected to line 1 of the sipura, the caller cannot hear 
anything.  I've only noticed this problem in this exact scenario.  The 
other situations listed below have no problems whatsoever and audio 
works in both directions:

1. Call from sipura line 1 to any internal SIP phone.
1. Call from any internal SIP phone to sipura line 1.
2. Call from sipura line 1 out through X100P.
3. Call into my X100P from outside and answer sipura line 2.
4. Call into my X100P from outside and answer sipura line 2 and THEN 
transfer to sipura line 1.
5. Call into my X100P from outside and answer sipura line 1 (the caller 
cannot hear audio for this leg of the conversation), TRANSFER to any 
other line, and transfer back to sipura line 1.  After the second 
transfer, the caller can hear audio from sipura line 1.

I don't know what is special about line 1.  I've switched my analog 
phones across the two ports on the sipura to make sure it wasn't one of 
my phones (not that I thought it was anyway).

Frankie, have you tried the same experiment, but pulled your analog 
phone from line 1 and put it in line 2?

Has anyone else seen issues like this with line 1 on a sipura?

Thanks..

-- Chris