app_dial.c: dial_exec_full Unable to create channel type SIP (cause 20 Unknown)

Question

I for the first time trying to configure the asterisk on My Ubuntu Linux Machine .I installed the asterisk and on executing the following command i am getting this in my terminal...

root@ubuntu:~# asterisk -r
Asterisk 1.8.4.4~dfsg-2ubuntu1.1, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
 =========================================================================
Connected to Asterisk 1.8.4.4~dfsg-2ubuntu1.1 currently running on ubuntu (pid = 1062)
ubuntu*CLI> 

Now i have updated sip.conf with following..

[general]
port = 5060
bindaddr = 0.0.0.0
context = others

[2000]
type=friend
context=my-phones
secret=1234
host=dynamic

[2001]
type=friend
context=my-phones
secret=1234
host=dynamic

And here is my extensions.conf

[others]

[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001) 

Now for testing purpose i have downloaded Twinkle softphone and tried to configure the extension 2000 with Username 2000 and Domain Localhost . Now i after registering above in twinkle i tried to call Extension 2001 from Twinkle but it is giving following error into the Asterisk CLI..

 [Apr  1 03:49:58] WARNING[2301]: app_dial.c:2041 dial_exec_full: Unable to create   channel of type 'SIP' (cause 20 - Unknown)

And here is Command sip show peers result in terminal..

  Name/username              Host                                    Dyn Forcerport ACL Port     Status     
  2000/2000                  127.0.0.1                                D          5061     Unmonitored 
  2001                       (Unspecified)                            D          0        Unmonitored 
  2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 1 offline]

Sip Registration Is also showing Zero..

ubuntu*CLI> sip show registry
Host                                    dnsmgr Username       Refresh    State                Reg.Time                 
0 SIP registrations.

This is all about my problem..I am very novice to asterisk and Linux both and have been struck badly into this situation. Please help me to resolve this.

Answer 1

• sip show registration have show zero
• sip show peer have show ip of your phone near peer 2001.

So you need register your softphone/phone on asterisk.

Considering level of your question I recommend you read asterisk book. I'd suggest you read ORelly's "Asterisk The Future of Telephony".

Answer 2

You are getting this error because 2001 is not registered anywhere.

Register 2001 in another softphone and try.

Answer 3

Step 1 - De register the subscriber 2001 from the softphone

Step 2 - Register again the user 2001, keeping up your asterisk server.

Answer 4

I know this is old, but for those researching this same issue:

@moonstruck provided the only correct answer.

OP configured 2 SIP extensions, but used only 1 SIP client (2000) to connect to asterisk.
Obviously, 2 SIP clients must be connected to asterisk before the call can be routed between them.

sip show registry lists registrations between SIP servers. It has nothing to do with SIP clients.

Answer 5

I have been searching the solution to this question since a few days now and could not find a single solution anywhere online. So i did my own research and found following interesting facts:-

FINDINGS:-

1. I have 13 extensions, all are part of every Queue i defined. At one time only 4 to 5 extensions are logged in (online).
2. Whenever some inbound call land on Queue, the Queue start searching for all 13 extensions. This is the time when Asterisk display this error.
3. At one time, asterisk display multiple errors. The total number of errors on one specific time could be defined as per following formula:

The Number of errors displayed at single time = No of total extensions in that Queue - No of online extensions

4. I tried this phenomena with different number of extensions and every time found the formula to be true.
5. The system has perfectly no issue in keep running smoothly with this error.

RECOMMENDATIONS:-

I would recommend to keep the system running and live with this error, unless someone find a cure for these annoying errors. Don't look at me; this is my first time on Asterisk and FreePBX. :)