Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Unknown)
Scott Huang wrote: Hi I've saw some similar case in the mail list, but seems no standard answers, so I decide ask here again. Is there anyone see the message below ? I use asterisk(1.8.11-cert 9) in my openbts2.8, and when I made a phone call, the Asterisk CLI poppd the following messages. = *CLI == Using SIP RTP CoS mark 5 -- Executing [8690@phones:1] Dial(SIP/IMSI466974600011287-, SIP/IMSI466974104638690) in new stack [Dec 18 16:00:49] WARNING[2934]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/IMSI466974600011287-' status is 'CHANUNAVAIL' == When you use a dynamic host type, the device needs to register to Asterisk in order to be dialed. Otherwise there is no way to for Asterisk to know what address to send the invite to and Asterisk will make chan_sip issue the cause 20 error you are seeing. If the device has a static IP and you don't want to deal with registration, you could always change the host to that IP address. Alternatively you could just figure out how to get your devices to register to your Asterisk server. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://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)
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
Re: [asterisk-users] unable to create channel of type 'SIP'
Can you give me some pointers on where to read documentation on how to set up registered phones? Also I'm wondering if maybe it would help if I tried setting up some softphones first. Can someone recommend some cheap softphones that work with asterisk? Jacob On Tue, May 29, 2012 at 5:36 PM, Danny Nicholas da...@debsinc.com wrote: You can dial out from an unregistered SIP peer, but you can't receive a call or call that peer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jacob Fenwick Sent: Tuesday, May 29, 2012 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] unable to create channel of type 'SIP' Good catch. Unfortunately, I actually did have it in there as dialGSM, I just copied from the wrong version of the file when I copied and pasted it here. This is what I get from sip show peers: Name/Username: IMSI262422146099205 Host: (Unspecified) Dyn: D Forceport: 0 ACL: Port: Unmonitored Status ... same for the other IMSI... 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] Jacob On Tue, May 29, 2012 at 5:25 PM, James Thomas jthomas...@gmail.com wrote: I think you need to change: exten = 2012,1,Macro(dialSIP,IMSI262428511722625) exten = 2013,1,Macro(dialSIP,IMSI262422146099205) to: exten = 2012,1,Macro(dialGSM,IMSI262428511722625) exten = 2013,1,Macro(dialGSM,IMSI262422146099205) also what does sip show peers show, as opposed to sip show registry? On Tue, May 29, 2012 at 2:55 PM, Jacob Fenwick jacob.fenw...@gmail.com wrote: I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAs terisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full: unable to create channel of type 'SIP' (cause 20 - unknown) If I type sip show registry it says there are 0 SIP registrations. Should I see both the phones registered at this point? If that's what's wrong, what am I doing wrong that's making the phones not able to register? Below is my Asterisk configuration. Jacob #/etc/asterisk/sip.conf [general] context=sip-external #... [IMSI262428511722625] callerid=2012 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic dtmfmode=info [IMSI262422146099205] callerid=2013 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic dtmfmode=info #/etc/asterisk/extensions.conf [macro-dialGSM] exten = s,1,Dial(SIP/${ARG1}) exten = s,2,Goto(s-${DIALSTATUS},1) exten = s-CANCEL,1,Hangup exten = s-NOANSWER,1,Hangup exten = s-BUSY,1,Busy(30) exten = s-CONGESTION,1,Congestion(30) exten = s-CHANUNAVAIL,1,playback(ss-noservice) exten = s-CANCEL,1,Hangup [sip-external] exten = 2012,1,Macro(dialSIP,IMSI262428511722625) exten = 2013,1,Macro(dialSIP,IMSI262422146099205) -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] unable to create channel of type 'SIP'
I think you need to change: exten = 2012,1,Macro(dialSIP,IMSI262428511722625) exten = 2013,1,Macro(dialSIP,IMSI262422146099205) to: exten = 2012,1,Macro(dialGSM,IMSI262428511722625) exten = 2013,1,Macro(dialGSM,IMSI262422146099205) also what does sip show peers show, as opposed to sip show registry? On Tue, May 29, 2012 at 2:55 PM, Jacob Fenwick jacob.fenw...@gmail.comwrote: I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full: unable to create channel of type 'SIP' (cause 20 - unknown) If I type sip show registry it says there are 0 SIP registrations. Should I see both the phones registered at this point? If that's what's wrong, what am I doing wrong that's making the phones not able to register? Below is my Asterisk configuration. Jacob #/etc/asterisk/sip.conf [general] context=sip-external #... [IMSI262428511722625] callerid=2012 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic dtmfmode=info [IMSI262422146099205] callerid=2013 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic dtmfmode=info #/etc/asterisk/extensions.conf [macro-dialGSM] exten = s,1,Dial(SIP/${ARG1}) exten = s,2,Goto(s-${DIALSTATUS},1) exten = s-CANCEL,1,Hangup exten = s-NOANSWER,1,Hangup exten = s-BUSY,1,Busy(30) exten = s-CONGESTION,1,Congestion(30) exten = s-CHANUNAVAIL,1,playback(ss-noservice) exten = s-CANCEL,1,Hangup [sip-external] exten = 2012,1,Macro(dialSIP,IMSI262428511722625) exten = 2013,1,Macro(dialSIP,IMSI262422146099205) -- _ -- 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 -- _ -- 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'
Good catch. Unfortunately, I actually did have it in there as dialGSM, I just copied from the wrong version of the file when I copied and pasted it here. This is what I get from sip show peers: Name/Username: IMSI262422146099205 Host: (Unspecified) Dyn: D Forceport: 0 ACL: Port: Unmonitored Status ... same for the other IMSI... 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] Jacob On Tue, May 29, 2012 at 5:25 PM, James Thomas jthomas...@gmail.com wrote: I think you need to change: exten = 2012,1,Macro(dialSIP,IMSI262428511722625) exten = 2013,1,Macro(dialSIP,IMSI262422146099205) to: exten = 2012,1,Macro(dialGSM,IMSI262428511722625) exten = 2013,1,Macro(dialGSM,IMSI262422146099205) also what does sip show peers show, as opposed to sip show registry? On Tue, May 29, 2012 at 2:55 PM, Jacob Fenwick jacob.fenw...@gmail.com wrote: I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full: unable to create channel of type 'SIP' (cause 20 - unknown) If I type sip show registry it says there are 0 SIP registrations. Should I see both the phones registered at this point? If that's what's wrong, what am I doing wrong that's making the phones not able to register? Below is my Asterisk configuration. Jacob #/etc/asterisk/sip.conf [general] context=sip-external #... [IMSI262428511722625] callerid=2012 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic dtmfmode=info [IMSI262422146099205] callerid=2013 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic dtmfmode=info #/etc/asterisk/extensions.conf [macro-dialGSM] exten = s,1,Dial(SIP/${ARG1}) exten = s,2,Goto(s-${DIALSTATUS},1) exten = s-CANCEL,1,Hangup exten = s-NOANSWER,1,Hangup exten = s-BUSY,1,Busy(30) exten = s-CONGESTION,1,Congestion(30) exten = s-CHANUNAVAIL,1,playback(ss-noservice) exten = s-CANCEL,1,Hangup [sip-external] exten = 2012,1,Macro(dialSIP,IMSI262428511722625) exten = 2013,1,Macro(dialSIP,IMSI262422146099205) -- _ -- 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 -- _ -- 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 -- _ -- 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'
You can dial out from an unregistered SIP peer, but you can't receive a call or call that peer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jacob Fenwick Sent: Tuesday, May 29, 2012 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] unable to create channel of type 'SIP' Good catch. Unfortunately, I actually did have it in there as dialGSM, I just copied from the wrong version of the file when I copied and pasted it here. This is what I get from sip show peers: Name/Username: IMSI262422146099205 Host: (Unspecified) Dyn: D Forceport: 0 ACL: Port: Unmonitored Status ... same for the other IMSI... 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] Jacob On Tue, May 29, 2012 at 5:25 PM, James Thomas jthomas...@gmail.com wrote: I think you need to change: exten = 2012,1,Macro(dialSIP,IMSI262428511722625) exten = 2013,1,Macro(dialSIP,IMSI262422146099205) to: exten = 2012,1,Macro(dialGSM,IMSI262428511722625) exten = 2013,1,Macro(dialGSM,IMSI262422146099205) also what does sip show peers show, as opposed to sip show registry? On Tue, May 29, 2012 at 2:55 PM, Jacob Fenwick jacob.fenw...@gmail.com wrote: I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAs terisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full: unable to create channel of type 'SIP' (cause 20 - unknown) If I type sip show registry it says there are 0 SIP registrations. Should I see both the phones registered at this point? If that's what's wrong, what am I doing wrong that's making the phones not able to register? Below is my Asterisk configuration. Jacob #/etc/asterisk/sip.conf [general] context=sip-external #... [IMSI262428511722625] callerid=2012 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic dtmfmode=info [IMSI262422146099205] callerid=2013 canreinvite=no type=friend context=sip-external allow=gsm host=dynamic dtmfmode=info #/etc/asterisk/extensions.conf [macro-dialGSM] exten = s,1,Dial(SIP/${ARG1}) exten = s,2,Goto(s-${DIALSTATUS},1) exten = s-CANCEL,1,Hangup exten = s-NOANSWER,1,Hangup exten = s-BUSY,1,Busy(30) exten = s-CONGESTION,1,Congestion(30) exten = s-CHANUNAVAIL,1,playback(ss-noservice) exten = s-CANCEL,1,Hangup [sip-external] exten = 2012,1,Macro(dialSIP,IMSI262428511722625) exten = 2013,1,Macro(dialSIP,IMSI262422146099205) -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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'
if sip show peers didn't work, probably the phone u trying to dial isn't registered as was said before. In sip.conf check the username, secret, host, userid. You made a mistake somewhere along the line. some phones will only register if host is set to dynamic...___ 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] Unable to create channel of type 'SIP'
On Thursday 21 April 2005 09:38, Ronald Wiplinger wrote: I (601) call one of my users (8862), after one minute I try to call him again and get Unable to create channel of type 'SIP' sip show peers does not list him. I cannot figure out why this happens and more important how I can fix it. You will get the Unable to create channel SIP if you do Dial(SIP/some-extension) and some-extension isn't registered with Asterisk. If sip show peers doesn't even show the 8862 entry, you should probably check you sip.conf. Anyway, sound to me like your SIP endpoint (8862) isn't re-registering before the initial registration times out or has problems with the sip.conf qualify option. I've seen a bunch of endpoints lose registration because qualify was set too low or because the endpoints didn't respond with something Asterisk could use for some reason. -- Regards, Tais M. Hansen ComX Networks A/S Tel: +45-70257474 Fax: +45-70257374 pgp9dl9fKymkh.pgp Description: PGP signature ___ 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] Unable to create channel of type 'SIP'
Hi I get the following error when i dial a sip extension, please help NOTICE[1681]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time The SIP extension you are trying to dial has not registered with *. T ___ 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] Unable to create channel of type 'SIP'
[EMAIL PROTECTED] wrote: I recently moved my Asterisk configuration to a new server and re-built Asterisk from CVS. Now, I'm experiencing the following issue with SIP: Executing Dial(Zap/1-1, SIP/100|20) in new stack NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to create channel of type 'SIP' == Everyone is busy at this time Has anyone seen this issue before? cvs update again Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users