Alexander Lopez
2005-Dec-25 16:14 UTC
[Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup
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@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > lee@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@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@lists.digium.com > > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > > > lee@michwave.com > > > Sent: Sunday, December 25, 2005 4:49 PM > > > To: asterisk-users@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 > > > > > > > > > ---------------------------------------------------------------- > > > 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 >
lee@michwave.com
2005-Dec-25 23:26 UTC
[Asterisk-Users] weird problem with sipura spa2000 and soundcardpa setup
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@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@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > > lee@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@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@lists.digium.com > > > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > > > > lee@michwave.com > > > > Sent: Sunday, December 25, 2005 4:49 PM > > > > To: asterisk-users@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 > > > > > > > > > > > > ---------------------------------------------------------------- > > > > 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.