[asterisk-bugs] [Asterisk 0017572]: app_dial.c: dial_exec_full Unable to create channel type 'SIP' (cause 20 - unknown)
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Jun 30 15:11:38 CDT 2010
The following issue has been UPDATED.
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https://issues.asterisk.org/view.php?id=17572
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Reported By: germ10
Assigned To:
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Project: Asterisk
Issue ID: 17572
Category: Applications/app_dial
Reproducibility: always
Severity: minor
Priority: normal
Status: closed
Asterisk Version: 1.4.31
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: no change required
Fixed in Version:
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Date Submitted: 2010-06-30 15:05 CDT
Last Modified: 2010-06-30 15:11 CDT
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Summary: app_dial.c: dial_exec_full Unable to create channel
type 'SIP' (cause 20 - unknown)
Description:
I am consistently getting this error in my Router Asterisk install.
app_dial.c: dial_exec_full Unable to create channel type 'SIP' (cause 20 -
unknown)
I am trying to test PC-to-PC Softphone. I have tried x-lite, idefisk, and
more.
I tried changing sip_local.conf & extensions_local.conf multiple times,
even with cut & paste just to make sure any typo errors.
Tried with qualify=yes/no. With or without username=
Disabled WinXp firewall on both machines, to no avail.
Could it be the default Asterisk install is missing something?
I did basic install from
http://ipkg.nslu2-linux.org/feeds/optware/ddwrt/cross/stable
Any info would be helpful?
Btw, I googled all over the place looking for answer.
I will upload Log & Config files if required.
Added #include sip_local.conf & #include extensions_local.conf to
corresponding files.
My sip_local.conf looks like,
[1000]
type=friend
secret=********
qualify=yes
nat=no
host=dynamic
context=from-internal
canreinvite=no
callerid=Toshiba <1000>
[1001]
type=friend
secret=********
qualify=yes
nat=no
host=dynamic
context=from-internal
canreinvite=no
callerid=Dell <1001>
My extensions_local.conf looks like,
[from-internal]
exten => 1000,1,Dial(SIP/1000)
exten => 1001,1,Dial(SIP/1001)
In second iteration (comments included), completly replaced above with,
[1000]
type=friend
secret=**************
;dtmfmode=rfc2833
callerid="First Phone" <1000>
host=dynamic ; The device must always register
canreinvite=no
; Deny registration from anywhere first
deny=0.0.0.0/0.0.0.0
; Replace the IP address and mask below with the actual IP address and
mask
; of the computer running the softphone, or the address of the hardware
phone,
; either a host address and full mask, or a network address and correct
mask,
; registering will be allowed from that host/network.
permit=192.168.1.112/255.255.255.0 ; current value as shown in connection
context=myphones
[1001]
type=friend
secret=**************
;dtmfmode=rfc2833
callerid="Second Phone" <1001>
host=dynamic ; The device must always register
canreinvite=no
; Deny registration from anywhere first
deny=0.0.0.0/0.0.0.0
; Replace the IP address and mask below with the actual IP address and
mask
; of the computer running the softphone, or the address of the hardware
phone,
; either a host address and full mask, or a network address and correct
mask,
; registering will be allowed from that host/network.
permit=192.168.1.114/255.255.255.0 ; current value as shown in connection
context=myphones
[myphones]
; When we dial something from the phones we just added in
; sip.conf, Asterisk will look for a matching extension here,
; in this context.
; First Phone, extension 1000. If 1000 is called, here is
; where we land, and the device registered with the
; name 1000, is dialed, after that Asterisk hangs up.
exten => 1000,1,Dial(SIP/1000)
exten => 1000,n,Hangup()
; The same goes for Second Phone, extension 1001
exten => 1001,1,Dial(SIP/1001)
exten => 1001,n,Hangup()
; Testing extension, prepare to be insulted like a
; Monthy Python knight
exten => 201,1,Answer()
exten => 201,n,Playback()
exten => 201,n,Hangup()
; Echo-test, it is good to test if we have sound in both directions.
; The call is answered
exten => 202,1,Answer()
; Welcome message is played
exten => 202,n,Playback()
; Play information about the echo test
exten => 202,n,Playback()
; Do the echo test, end with the # key
exten => 202,n,Echo()
; Plays information that the echo test is done
exten => 202,n,Playback()
; Goodbye message is played
exten => 202,n,Playback()
; Hangup() ends the call, hangs up the line
exten => 202,n,Hangup()
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----------------------------------------------------------------------
(0124113) pabelanger (manager) - 2010-06-30 15:11
https://issues.asterisk.org/view.php?id=17572#c124113
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Thanks for your comments. This does not appear to be a bug report and we
are closing it. We appreciate the difficulties you are facing, but it would
make more sense to raise your question in the support tracker,
http://www.asterisk.org/support
Issue History
Date Modified Username Field Change
======================================================================
2010-06-30 15:11 pabelanger Note Added: 0124113
2010-06-30 15:11 pabelanger Status new => closed
2010-06-30 15:11 pabelanger Resolution open => no change
required
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