[asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Hi, I'm using Cicso - Linksys SPA3102 to connect to asterisk. I have followed the official user manual and the blog post here http://www.skelleton.net/2012/08/02/linksys-spa-3102/ When I call an extension say 225 from the analog phone, I can get the IVR I have setup in my dialplan. But when I Call the analog phone extension using a sip phone I get the following error message: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) If any more information is required, i'd be glad to post it here. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Face wrote: Well, thanks for responding. I went back to 10.10.0 and things seem to be working fine now! This is certainly good to know but I'd like to know why upgrading to 11 did not seem to work for you. This is the first case since it's been out where it doesn't appear to have been smooth. Would you be willing to provide the information I asked about from a running 11 instance? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
I upgrading to 11 because I want to use the MessageSend command from the AMI, ver 10 dose not have MessageSend In the list of commands. Unfortunately I remove ver 11 and I dont think I can provide the information you asked. On Tue, Nov 20, 2012 at 4:46 PM, Joshua Colp jc...@digium.com wrote: Face wrote: Well, thanks for responding. I went back to 10.10.0 and things seem to be working fine now! This is certainly good to know but I'd like to know why upgrading to 11 did not seem to work for you. This is the first case since it's been out where it doesn't appear to have been smooth. Would you be willing to provide the information I asked about from a running 11 instance? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Sincerely, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Face wrote: Hello, Hola, After Upgrade to Asterisk 11.1.0-rc1 I keep getting == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [603@DLPN_AlDimnaDialPlan:601] Dial(SIP/601-0002, SIP/603) in new stack [Nov 16 06:42:33] WARNING[15547][C-0004]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/601-0002' status is 'CHANUNAVAIL' and would not go to voicemail? Unfortunately without more information (dialplan involved, complete console output, sip show peer 603) it's impossible to fathom any potential reason why this is occurring. I suspect that's why nobody has responded to you until now. If you can provide that information I'm sure we can all help to determine if there really is an issue at work here! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
On Mon, Nov 19, 2012 at 3:51 PM, Joshua Colp jc...@digium.com wrote: Face wrote: Hello, Hola, After Upgrade to Asterisk 11.1.0-rc1 I keep getting == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [603@DLPN_AlDimnaDialPlan:601] Dial(SIP/601-0002, SIP/603) in new stack [Nov 16 06:42:33] WARNING[15547][C-0004]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/601-0002' status is 'CHANUNAVAIL' and would not go to voicemail? Unfortunately without more information (dialplan involved, complete console output, sip show peer 603) it's impossible to fathom any potential reason why this is occurring. I suspect that's why nobody has responded to you until now. If you can provide that information I'm sure we can all help to determine if there really is an issue at work here! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well, thanks for responding. I went back to 10.10.0 and things seem to be working fine now! -- Sincerely, falazemi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Hello, After Upgrade to Asterisk 11.1.0-rc1 I keep getting == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [603@DLPN_AlDimnaDialPlan:601] Dial(SIP/601-0002, SIP/603) in new stack [Nov 16 06:42:33] WARNING[15547][C-0004]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/601-0002' status is 'CHANUNAVAIL' and would not go to voicemail? -- Sincerely, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
