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

Alexander Lopez alex.lopez at opsys.com
Sun Dec 25 16:14:08 MST 2005


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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> 
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