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Summary:ASTERISK-16305: app_dial.c: dial_exec_full Unable to create channel type 'SIP' (cause 20 - unknown)
Reporter:Albert Costa (germ10)Labels:
Date Opened:2010-06-30 15:05:34Date Closed:2011-06-07 14:04:48
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Applications/app_dial
Versions:Frequency of
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Environment:Attachments:
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()

Comments:By: Paul Belanger (pabelanger) 2010-06-30 15:11:37

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