[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
[asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Unknown)
2012/12/19 Scott Huang gyration.hu...@gmail.com 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' == The attached files are the sip.conf and extension.conf and wireshark trace log. The part of my setting in sip.conf is: [IMSI466974104638690]; callerid=8690 8690 ; regexten=8690; canreinvite=no type=friend allow=gsm context=phones host=dynamic registertrying=yes [IMSI466974102820333]; callerid=0333 0333 ; regexten=0333; canreinvite=no type=friend allow=gsm context=phones host=dynamic registertrying=yes [IMSI466974600011287]; callerid=1287 1287 ; regexten=1287; canreinvite=no type=friend allow=gsm context=phones host=dynamic registertrying=yes The part of my setting in extensions.conf is: [phones] exten = 8690,1,Dial(SIP/IMSI466974104638690) exten = 0333,1,Dial(SIP/IMSI466974102820333) exten = 1287,1,Dial(SIP/IMSI466974600011287) How to exactly configure asterisk for a sip call ? Thanks very much ! BR/Scott -- _ -- 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 - 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
[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
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)
--
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asterisk-users
[asterisk-users] unable to create channel of type 'SIP'
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)
--
_
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asterisk-users mailing list
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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)
--
_
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asterisk-users mailing list
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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)
--
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http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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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)
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[Asterisk-Users] Unable to create channel of type 'SIP'
Hi, i have a working asterisk version CVS-v1-0-05/13/05-15:06:32, i was installed using amportal , i want to migrate to another server, this time i dont wat to use amportal and edited by hand everyfile, i can make outboundcalls without problems, but i cant receive anything, either from between the sip phones or the external peers, i copied the sip.conf from the old server, this is the relevant port of the external peer, a cisco as5400: ### sip.conf ### [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=g729 allow=g723.1 allow=alaw allow=ulaw context = bogon-calls ; Send unknown SIP callers to this context callerid = Unknown language=es register = @prepago-in [prepago-in] type=friend host=aaa.bbb.ccc.ddd context = from-external dtmfmode=rfc2833 insecure=very ; required for incoming FWD calls [prepago-out] type=peer ; we only want to call out, not be called host=aaa.bbb.ccc.ddd dtmfmode=rfc2833 [22662124] callerid=22662124 22662124 context=from-internal host=dynamic secret=22662124 type=friend username=22662124 this is the error log Destroying call '[EMAIL PROTECTED]' Feb 23 19:16:52 NOTICE[1023]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Transmitting (no NAT) to aaa.bbb.ccc.ddd:5060: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP aaa.bbb.ccc.ddd:5060;received=aaa.bbb.ccc.ddd From: sip:[EMAIL PROTECTED];tag=1FA0C538-A3B To: sip:[EMAIL PROTECTED];tag=as3aaf6cd5 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 X-Asterisk-HangupCause: No route to destination what can we wrong? --- Miguel ___ --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
[Asterisk-Users] Unable to create channel of type SIP-please help
Hi there,
Im having a hard time getting outbound calling
to my SIPPSTN gateway. I
continuasly get the following result in my log files:
Jun 5 10:07:50 WARNING[1568]: No such host: t2y
Jun 5 10:07:50 NOTICE[1568]: Unable to create channel of type 'SIP'
Jun 5 10:07:50 VERBOSE[1568]: == Everyone is busy/congested at this time
I make the following context in my extensions.conf
file
exten = s,1,SetCallerID(${T2Y1})
exten = s,2,SetCIDName(${MYNAME})
exten = s,3,Dial(SIP/[EMAIL PROTECTED],${ARG2})
exten = s,4,Playback(new/acnt-or-cir-busy-now)
exten = s,5,Hangup
I do have a [t2y] reference in my SIP.conf file
[t2y]
type=friend
secret=PASSWORD
username=USERNAME
host=budgetphone.nl
dtmfmode=rfc2833
fromuser= USERNAME
fromdomain=budgetphone.nl
username= USERNAME
insecure=very
nat=yes
qualify=no
register=yes
I tried changing to change the reference t2y in the
context in extensions.conf to @budgetphone.nl and @sip.budgetphone.nl, but then
I get a authorization error.
Jun 5 10:16:24 WARNING[1568]: Forbidden - wrong
password on authentication for INVITE to 'Guy Soudant ;tag=as27b52309'
Jun 5 10:16:24 VERBOSE[1568]: -- SIP/sip.budgetphone.nl-adeb is circuit-busy
Im lost here. Could someone help me out?
Thanks
Guy
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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
[Asterisk-Users] Unable to create channel of type 'SIP'
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. -- Executing Dial(SIP/601-0f22, SIP/8862|60|tr) in new stack Apr 21 15:30:47 NOTICE[14706]: app_dial.c:973 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) -- Executing NoOp(SIP/601-0f22, CONGESTION) in new stack -- Executing VoiceMail(SIP/601-0f22, u8862) in new stack -- Playing '/var/spool/asterisk/voicemail/others/8862/unavail' (language 'en') -- Playing 'vm-intro' (language 'en') == Spawn extension (VoIP_Phone, 8862, 3) exited non-zero on 'SIP/601-0f22' bye Ronald ___ 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
[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 ___ 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
[Asterisk-Users] Unable to create channel of type 'SIP'
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? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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
