When calling through Plainvoip from my Asterisk at Home box I get the following log entries. Jun 8 01:27:25 VERBOSE[5550] logger.c: -- Called plainvoip/1<insert #> Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Call accepted by 66.199.240.2 (format g729) Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729 Jun 8 01:27:26 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 is ringing Jun 8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 stopped sounds Jun 8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 answered SIP/503-6d4c Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to 256 What I hear on the phone is one ring and then nothing. This has only been in the past few days. Has anybody else had a problem like this? -- Henry J. Cobb http://www.io.com/~hcobb/
Do you have the g729 codec? On 6/8/06, Henry J. Cobb <hcobb@io.com> wrote:> When calling through Plainvoip from my Asterisk at Home box I get the > following log entries. > > Jun 8 01:27:25 VERBOSE[5550] logger.c: -- Called plainvoip/1<insert #> > Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Call accepted by > 66.199.240.2 (format g729) > Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729 > Jun 8 01:27:26 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 is ringing > Jun 8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 stopped > sounds > Jun 8 01:27:29 VERBOSE[5550] logger.c: -- IAX2/plainvoip-3 answered > SIP/503-6d4c > Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to 256 > > What I hear on the phone is one ring and then nothing. > > This has only been in the past few days. > > Has anybody else had a problem like this? > > -- > Henry J. Cobb > http://www.io.com/~hcobb/ > > _______________________________________________ > --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 >-- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856
> Do you have the g729 codec? > > On 6/8/06, Henry J. Cobb <hcobb@io.com> wrote: >> Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729...>> Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to >> 256Yes, and that works fine when talking with the phone itself, as you see the connection to the phone is g729. Then it changes from g729 to g729? -- Henry J. Cobb http://www.io.com/~hcobb/
Send an email to support@plainvoip.com. They are normally quite helpful. bp On 6/8/06, Henry J. Cobb <hcobb@io.com> wrote:> > > Do you have the g729 codec? > > > > On 6/8/06, Henry J. Cobb <hcobb@io.com> wrote: > >> Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is g729 > ... > >> Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to > >> 256 > > Yes, and that works fine when talking with the phone itself, as you see > the connection to the phone is g729. > > Then it changes from g729 to g729? > > -- > Henry J. Cobb > http://www.io.com/~hcobb/ > > _______________________________________________ > --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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060608/d489d9c2/attachment.htm
could it be IPP VS digium implementation ? On 6/8/06, William Piper <william.piper@gmail.com> wrote:> > Send an email to support@plainvoip.com. They are normally quite helpful. > > bp > > > On 6/8/06, Henry J. Cobb <hcobb@io.com> wrote: > > > > > Do you have the g729 codec? > > > > > > On 6/8/06, Henry J. Cobb < hcobb@io.com> wrote: > > >> Jun 8 01:27:26 VERBOSE[2798] logger.c: -- Format for call is > > g729 > > ... > > >> Jun 8 01:27:29 DEBUG[2798] chan_iax2.c: Ooh, voice format changed to > > >> 256 > > > > Yes, and that works fine when talking with the phone itself, as you see > > the connection to the phone is g729. > > > > Then it changes from g729 to g729? > > > > -- > > Henry J. Cobb > > http://www.io.com/~hcobb/ <http://www.io.com/%7Ehcobb/> > > > > _______________________________________________ > > --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 > > >-- Mike Sales Manager http://www.theclubvoip.com Making it happen 1.888.470.7253 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060609/0f58528b/attachment.htm
"Mike Lynchfield" <theclubvoip@gmail.com> wrote:> could it be IPP VS digium implementation ?Actually it seems to have been a NAT issue and adding canreinvite=no (as suggested by another person offlist) fixed it up. -- Henry J. Cobb http://www.io.com/~hcobb/