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 <[EMAIL PROTECTED]>: > 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: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > [EMAIL PROTECTED] > > 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 <[EMAIL PROTECTED]>: > > > > > 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: [EMAIL PROTECTED] > > > > [mailto:[EMAIL PROTECTED] On Behalf Of > > > > [EMAIL PROTECTED] > > > > Sent: Sunday, December 25, 2005 4:49 PM > > > > To: [email protected] > > > > 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 > > > > > > > > > > > > ---------------------------------------------------------------- > > > > This message was sent using IMP, the Internet Messaging Program. > > > > > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > > > Asterisk-Users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > ---------------------------------------------------------------- > > This message was sent using IMP, the Internet Messaging Program. > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
