[Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup

[lee at michwave.com]

Yup all ata's can talk to each other just fine.  I can call one for another,
they all can make out going calls, and all receive phone call just fine

sip.conf
-----------------------------------------------
sipura
------
[sipura1-1]
type=friend
username=<username>
secret=<password>
host=dynamic
nat=no
callerid="name" <999-999-9999>
reinvite=no
canreinvite=no
context=localphone
qualify=yes
callgroup=1
pickupgroup=1
disallow=all
allow=ulaw

cisco ATA
---------
[leesata]
type=friend
username=<name>
secret=<password>
host=dynamic
nat=no
callerid="name2" <888-888-8888>
canreinvite=no
context=localphone
qualify=yes

and yes alsa.conf file has context=localphone also

-------------------------------------------------------------
as for debugging, The error below is all I get no matter what debug level I run

-Lee 


Quoting Alexander Lopez <alex.lopez at opsys.com>:

> I don't know what codec the console is set to if any actualy since
> Astersk would do thje ttranscoding. It may even be signed linear, (don't
> quote me on that!!)
> 
> Can the Sipuras and Cisco talk to each other?? 
> How are the Phones set up in Sip.conf?
> Can you set debug to more detail?? (asterisk
> -rvvvvvvvvvvvvvvvvvvvvvvvvvv)
> 
> 
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com 
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> > lee at michwave.com
> > Sent: Sunday, December 25, 2005 5:19 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [Asterisk-Users] weird problem with sipura 
> > spa2000 and soundcardpa setup
> > 
> > I have my sipura set to a preferred codec of G711u but I also 
> > have it set to use any codec. The list of codecs are G711u G711a
> > G726-16
> > G726-24
> > G726-32
> > G726-40
> > G729a
> > G723
> > 
> > Is there a place to set the codec to use on the console 
> > device that I am missing.  There is nothing listed in the 
> > alsa.conf file
> > 
> > -Lee
> > 
> > 
> > Quoting Alexander Lopez <alex.lopez at opsys.com>:
> > 
> > > It is posible that your SPA is trying to use a codec that is not 
> > > available. I can't tell from the errors you provided.
> > > 
> > > Double check what codecs the Cisco is using and set the Spa to thwe 
> > > same....
> > > 
> > > Alex
> > >  
> > > 
> > > > -----Original Message-----
> > > > From: asterisk-users-bounces at lists.digium.com
> > > > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> > > > lee at michwave.com
> > > > Sent: Sunday, December 25, 2005 4:49 PM
> > > > To: asterisk-users at lists.digium.com
> > > > Subject: [Asterisk-Users] weird problem with sipura spa2000 and 
> > > > sound cardpa setup
> > > > 
> > > > Hello,
> > > >      Just joined this list in hopes of getting an answer to my 
> > > > problem and helping others in the future.  Anyways here is my 
> > > > problem
> > > > 
> > > > 
> > > >      I have asterisk 1.2.1 installed and setup the onboad 
> > sound card 
> > > > to autoanswer in the alsa.conf file to act as a pa system.  I 
> > > > currently have the extention setup to 66 to dial the sound card
> > > > 
> > > > exten => 66,1,Dial(Console/dsp)
> > > > 
> > > > If I dial it using my 7940 cisco phone, it works just fine.  
> > > > If I dial it using a cisco ata 186, it works just fine.  
> > If i dial 
> > > > from a phone connected to a sipura spa-2000 i get the following 
> > > > error.
> > > > 
> > > > --------------------------------------------------------------
> > > > ---------------
> > > > 
> > > >     -- Executing Dial("SIP/sipura1-2-bbb8", "Console/dsp") in new 
> > > > stack  << Call placed to 'dsp' on console >>  << Auto-answered >>
> > > >     -- Called dsp
> > > >     -- ALSA/default answered SIP/sipura1-2-bbb8 Dec 26
> > > > 04:55:14 ERROR[7332]: chan_alsa.c:643 alsa_write: Write
> > > > error: Unknown error 170  << Hangup on console >>
> > > >   == Spawn extension (localphone, 66, 1) exited non-zero on 
> > > > 'SIP/sipura1-2-bbb8'
> > > > 
> > > > --------------------------------------------------------------
> > > > ---------------
> > > > 
> > > > This leads me to believe I need to change a setting on the sipura 
> > > > for it must be sending something asterisk doesn't like.  
> > Other then 
> > > > this error, the sipura works fine.  I can make and 
> > receive calls on 
> > > > it just fine thru either a true voip connection or with 
> > my hard line 
> > > > with a x100p card.  I have tried dialing the soundcard with 2 
> > > > different sipura spa2000 and i get the same error with both.  
> > > > Anybody else run into this problem?
> > > > 
> > > > 
> > > > -Lee
> > > > 
> > > > 
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