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.<br>
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<br><br><div><span class="gmail_quote">On 12/26/05, <b class="gmail_sendername"><a href="mailto:lee@michwave.com">lee@michwave.com</a></b> <<a href="mailto:lee@michwave.com">lee@michwave.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Yup all ata's can talk to each other just fine. I can call one for another,<br>they all can make out going calls, and all receive phone call just fine<br><br>sip.conf<br>-----------------------------------------------<br>
sipura<br>------<br>[sipura1-1]<br>type=friend<br>username=<username><br>secret=<password><br>host=dynamic<br>nat=no<br>callerid="name" <999-999-9999><br>reinvite=no<br>canreinvite=no<br>context=localphone
<br>qualify=yes<br>callgroup=1<br>pickupgroup=1<br>disallow=all<br>allow=ulaw<br><br>cisco ATA<br>---------<br>[leesata]<br>type=friend<br>username=<name><br>secret=<password><br>host=dynamic<br>nat=no<br>callerid="name2" <888-888-8888>
<br>canreinvite=no<br>context=localphone<br>qualify=yes<br><br>and yes alsa.conf file has context=localphone also<br><br>-------------------------------------------------------------<br>as for debugging, The error below is all I get no matter what debug level I run
<br><br>-Lee<br><br><br>Quoting Alexander Lopez <<a href="mailto:alex.lopez@opsys.com">alex.lopez@opsys.com</a>>:<br><br>> I don't know what codec the console is set to if any actualy since<br>> Astersk would do thje ttranscoding. It may even be signed linear, (don't
<br>> quote me on that!!)<br>><br>> Can the Sipuras and Cisco talk to each other??<br>> How are the Phones set up in Sip.conf?<br>> Can you set debug to more detail?? (asterisk<br>> -rvvvvvvvvvvvvvvvvvvvvvvvvvv)
<br>><br>><br>> > -----Original Message-----<br>> > From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>> > [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">
asterisk-users-bounces@lists.digium.com</a>] On Behalf Of<br>> > <a href="mailto:lee@michwave.com">lee@michwave.com</a><br>> > Sent: Sunday, December 25, 2005 5:19 PM<br>> > To: Asterisk Users Mailing List - Non-Commercial Discussion
<br>> > Subject: RE: [Asterisk-Users] weird problem with sipura<br>> > spa2000 and soundcardpa setup<br>> ><br>> > I have my sipura set to a preferred codec of G711u but I also<br>> > have it set to use any codec. The list of codecs are G711u G711a
<br>> > G726-16<br>> > G726-24<br>> > G726-32<br>> > G726-40<br>> > G729a<br>> > G723<br>> ><br>> > Is there a place to set the codec to use on the console<br>> > device that I am missing. There is nothing listed in the
<br>> > alsa.conf file<br>> ><br>> > -Lee<br>> ><br>> ><br>> > Quoting Alexander Lopez <<a href="mailto:alex.lopez@opsys.com">alex.lopez@opsys.com</a>>:<br>> ><br>> > > It is posible that your SPA is trying to use a codec that is not
<br>> > > available. I can't tell from the errors you provided.<br>> > ><br>> > > Double check what codecs the Cisco is using and set the Spa to thwe<br>> > > same....<br>> > >
<br>> > > Alex<br>> > ><br>> > ><br>> > > > -----Original Message-----<br>> > > > From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com
</a><br>> > > > [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of<br>> > > > <a href="mailto:lee@michwave.com">lee@michwave.com
</a><br>> > > > Sent: Sunday, December 25, 2005 4:49 PM<br>> > > > To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>> > > > Subject: [Asterisk-Users] weird problem with sipura spa2000 and
<br>> > > > sound cardpa setup<br>> > > ><br>> > > > Hello,<br>> > > > Just joined this list in hopes of getting an answer to my<br>> > > > problem and helping others in the future. Anyways here is my
<br>> > > > problem<br>> > > ><br>> > > ><br>> > > > I have asterisk 1.2.1 installed and setup the onboad<br>> > sound card<br>> > > > to autoanswer in the
alsa.conf file to act as a pa system. I<br>> > > > currently have the extention setup to 66 to dial the sound card<br>> > > ><br>> > > > exten => 66,1,Dial(Console/dsp)<br>> > > >
<br>> > > > If I dial it using my 7940 cisco phone, it works just fine.<br>> > > > If I dial it using a cisco ata 186, it works just fine.<br>> > If i dial<br>> > > > from a phone connected to a sipura spa-2000 i get the following
<br>> > > > error.<br>> > > ><br>> > > > --------------------------------------------------------------<br>> > > > ---------------<br>> > > ><br>> > > > -- Executing Dial("SIP/sipura1-2-bbb8", "Console/dsp") in new
<br>>
> > > stack << Call placed to 'dsp' on
console >> << Auto-answered >><br>> > > > -- Called dsp<br>> > > > -- ALSA/default answered SIP/sipura1-2-bbb8 Dec 26<br>> > > > 04:55:14 ERROR[7332]: chan_alsa.c:643 alsa_write: Write
<br>> > > > error: Unknown error 170 << Hangup on console >><br>> > > > == Spawn extension (localphone, 66, 1) exited non-zero on<br>> > > > 'SIP/sipura1-2-bbb8'<br>> > > >
<br>> > > > --------------------------------------------------------------<br>> > > > ---------------<br>> > > ><br>> > > > This leads me to believe I need to change a setting on the sipura
<br>> > > > for it must be sending something asterisk doesn't like.<br>> > Other then<br>> > > > this error, the sipura works fine. I can make and<br>> > receive calls on<br>> > > > it just fine thru either a true voip connection or with
<br>> > my hard line<br>> > > > with a x100p card. I have tried dialing the soundcard with 2<br>> > > > different sipura spa2000 and i get the same error with both.<br>> > > > Anybody else run into this problem?
<br>> > > ><br>> > > ><br>> > > > -Lee<br>> > > ><br>> > > ><br>> > > > ----------------------------------------------------------------<br>> > > > This message was sent using IMP, the Internet Messaging Program.
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