Re: [asterisk-users] Transfer fails

2010-07-02 Thread Jonas Kellens 

Danny,

thank you for you feedback.

I have the following setting in sip.conf :

limitonpeer = yes

and for every sip peer definition I have :

asterisk*CLI> sip show peer test1

  * Name   : test1
  Realtime peer: Yes, cached
  Secret   : 
  MD5Secret: 
  Context  : from-TEST
  Subscr.Cont. : 

  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 4



With a call limit of 4, I think it must be possible to transfer a call, 
no ?!



Jonas.


On 07/02/2010 03:02 PM, Danny Nicholas wrote:


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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Re: [asterisk-users] Transfer fails

2010-07-02 Thread Danny Nicholas 
 

 

  _  

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 :


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


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/test1-010e", "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/test1-010c", "SIP/test2") in new stack
[Jul  2 10:55:02] -- Called test2
[Jul  2 10:55:02] -- SIP/test2-010d 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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