Debug info and posting your .confs will help to get replys.> -----Original Message----- > From: Paul [mailto:digium-list@9ux.com] > Sent: Sunday, November 27, 2005 9:52 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Voicepulse Open Access status? > > I have not had my 2 voicepulse open access numbers work for about 3 > weeks now. I have even tried to make them work on a new server build > using asterisk 1.2 > > I have several SIP and IAX DID's working fine. They worked fine on > asterisk 1.0.x and easily were moved to the 1.2 setup. > > sip show registry indicates they are registered oaky but calls to both > numbers go to voicepulse voice mail. If I setup hunt and fileters atthe> voicepulse web portal, that seems to work. For example, I can make the > numbers ring my cell phone instead of going to voice mail. The primary > number on the account works with the SPA-2000 ATA fine. I just used it > to call the vonage number that comes into the asterisk system and I am > using echo test as I type this. > > If anyone here has working voicepulse open access numbers could you > please post sample lines from sip.conf and extensions.conf? If anyone > here has been experiencing the same type of extended outage, I would > like to hear about it because I am going to ask them to waive charges > for at least one month of these softphone numbers. > > I was very tempted to put "Please help!!!" in the subject line today. > > If I don't get any replies I guess that means voicepulse sucks and I > should cancel these numbers :) > > _______________________________________________ > --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
I thought it might make more sense to start with an example config from someone who currently has incoming calls working. As I mentioned already sip show registry lists the 2 voicepulse numbers as registered. I have a console open with -rvvvvv and get no messages when I dial the numbers. In the past I have always been able to get some diagnostic info on the console when registered if something like context or codecs was amiss. Steve Totaro wrote:>Debug info and posting your .confs will help to get replys. > > > >>-----Original Message----- >>From: Paul [mailto:digium-list@9ux.com] >>Sent: Sunday, November 27, 2005 9:52 AM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: [Asterisk-Users] Voicepulse Open Access status? >> >>I have not had my 2 voicepulse open access numbers work for about 3 >>weeks now. I have even tried to make them work on a new server build >>using asterisk 1.2 >> >>I have several SIP and IAX DID's working fine. They worked fine on >>asterisk 1.0.x and easily were moved to the 1.2 setup. >> >>sip show registry indicates they are registered oaky but calls to both >>numbers go to voicepulse voice mail. If I setup hunt and fileters at >> >> >the > > >>voicepulse web portal, that seems to work. For example, I can make the >>numbers ring my cell phone instead of going to voice mail. The primary >>number on the account works with the SPA-2000 ATA fine. I just used it >>to call the vonage number that comes into the asterisk system and I am >>using echo test as I type this. >> >>If anyone here has working voicepulse open access numbers could you >>please post sample lines from sip.conf and extensions.conf? If anyone >>here has been experiencing the same type of extended outage, I would >>like to hear about it because I am going to ask them to waive charges >>for at least one month of these softphone numbers. >> >>I was very tempted to put "Please help!!!" in the subject line today. >> >>If I don't get any replies I guess that means voicepulse sucks and I >>should cancel these numbers :) >> >>_______________________________________________ >>--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 > >
> > > > I thought it might make more sense to start with an example configfrom> > someone who currently has incoming calls working. > > > > As I mentioned already sip show registry lists the 2 voicepulsenumbers> > as registered. I have a console open with -rvvvvv and get nomessages> > when I dial the numbers. In the past I have always been able to getsome> > diagnostic info on the console when registered if something likecontext> > or codecs was amiss. > > At the CLI, type 'sip debug' and call the numbers again. There shouldbe> something in the debug messages that point to the problem. Post the > results. >IAX debug as well.