_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 [...@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 [...@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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