[asterisk-users] Transfer fails

[Danny Nicholas]

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, July 02, 2010 4:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Transfer fails

 

Hello list,

this is the dialplan :

<snip>
exten => s,n,Dial(SIP/test1&SIP/test2,,t)
<snip>

exten => 10,1,Dial(SIP/test1)
exten => 20,1,Dial(SIP/test2)


So there is an incoming call that rings SIPaccounts test1 and test2.
Account test1 answers and wants to transfer the call to test2.
Transfer is : #20

This is what the CLI shows :

[Jul  2 10:55:30]     -- Executing [20 at from-TEST:1]
Dial("SIP/test1-0000010e", "SIP/test2") in new stack
[Jul  2 10:55:30] WARNING[7604]: app_dial.c:1296 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
[Jul  2 10:55:30]   == Everyone is busy/congested at this time (1:0/0/1)

...and the call is disconnected.

When I call the extension 20 directly from SIPaccount test1, the CLI shows
no problem :

[Jul  2 10:55:02]     -- Executing [20 at from-TEST:1]
Dial("SIP/test1-0000010c", "SIP/test2") in new stack
[Jul  2 10:55:02]     -- Called test2
[Jul  2 10:55:02]     -- SIP/test2-0000010d is ringing


So why can I call extension 20 (test2) directly but not transfer a call to
it ??


Jonas.

 

-- 

A good possibility is that you have an over-restrictive call-limit (or
whatever it's called in your branch) that is "filling the bucket" on the
incoming call and not allowing a transfer.

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