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

[Jorge Cisneros]

Hi, check in the sipura in advanced mode the parameter of RTP Packet Size
change it to 0.020 maybe with this you can fix the problem.






On 12/26/05, lee at michwave.com <lee at michwave.com> wrote:
>
> 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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