Re: [Asterisk-Users] Dial to a SIP fone ends up at Voicemail Busy
Hi, Terry H. Gilsenan wrote: I was having this problem with Gradstream BT101's with Asterisk @ Home version 0.7. The problem was that there was a sip channel still open (as far as asterisk and the phone were concerned) however this sip channel was not actually in use. The existence of this sip channel meant that whilst the phone could make calls, any incoming calls were directed to voicemail. Thanks for the hint. I did control the channels, they were all closed but the problem was still there. After testing the meeting app though (calling in via a PSTN-Cisco-Asterisk there is indeed a hung channel. Anyone knows what could be causing this? -- Channel (ContextExtensionPri ) State Appl. Data Zap/pseudo-1655835607 (defaults1 ) Rsrvd (None) (None) SIP/x.x.x.x-0814dbb8 (la-in 310 2 ) Up MeetMe |ip 2 active channel(s) -- cheers, Arnd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial to a SIP fone ends up at Voicemail Busy
C F wrote: are these phones behind nat? Yes, but correctly registered. The same fones dont have any problems when registered to a SER Server. Can constantly reloading the configuration cause problems? cheers, Arnd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial to a SIP fone ends up at Voicemail Busy
Hi, I was having this problem with Gradstream BT101's with Asterisk @ Home version 0.7. The problem was that there was a sip channel still open (as far as asterisk and the phone were concerned) however this sip channel was not actually in use. The existence of this sip channel meant that whilst the phone could make calls, any incoming calls were directed to voicemail. At the time what we had to to was to kill those persistent sip channels. I have not upgraded to Asterisk @ home version 1.0 and the problem no longer occurs. What version of Asterisk are you using? Connect to the CLI and issue the commands sip show channels zap show channels Are there any channels showing there that you know should be down? Regards, T -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arnd Vehling Sent: Wednesday, 25 May 2005 6:28 PM To: C F; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dial to a SIP fone ends up at Voicemail Busy C F wrote: are these phones behind nat? Yes, but correctly registered. The same fones dont have any problems when registered to a SER Server. Can constantly reloading the configuration cause problems? cheers, Arnd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial to a SIP fone ends up at Voicemail Busy
Hi, some of my sip fones which have several external numbers assigned are not reachable after a certain timespan. Instead of the fone the Voicemailbox is trigger in busy mode. After a reboot if the sip-fone the problem goes away for some time. Ive seen this problem with Sipuras and Grandstreams. Any ideas? cheers, Arnd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial to a SIP fone ends up at Voicemail Busy
are these phones behind nat? On 5/24/05, Arnd Vehling [EMAIL PROTECTED] wrote: Hi, some of my sip fones which have several external numbers assigned are not reachable after a certain timespan. Instead of the fone the Voicemailbox is trigger in busy mode. After a reboot if the sip-fone the problem goes away for some time. Ive seen this problem with Sipuras and Grandstreams. Any ideas? cheers, Arnd ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
