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 15:05:34 | Date Closed: | 2011-06-07 14:04:48 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_dial |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
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 |