[asterisk-users] Supervised call transfer problem

[Midiclorian]

Hi all,
   
  I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it)
   
  If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at weboffice.dyndns.org), the attendand (or supervised) transfer fails while the unattendandt (or direct) transfer works.
   
  Basically the errore I have is
  Supervised transfer requested, but unable to find callid 'f1d4a57c-f11e291e-22ed860d at 2.168.1.230'. Both legs must reside on Asterisk box to transfer at this time.
   
  Is there a way to allow the Transfer also for legs that are residing on Asterisk box?
   
  Here there is the console log of the supervised transfer failure.
   
  [root at webofficex asterisk]# asterisk -r
  Asterisk 1.2.24, Copyright (C) 1999 - 2007 Digium, Inc. and others.
  Created by Mark Spencer <markster at digium.com>
  Asterisk comes with ABSOLUTELY NO WARRANTY; type '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 'show license' for details.
  =========================================================================
  Connected to Asterisk 1.2.24 currently running on webofficex (pid = 27166)
  Verbosity is at least 3
  -- Executing Dial("SIP/390641043101-092ee290", "SIP/530 at weboffice.dyndns.org") in new stack
  -- Called 530 at weboffice.dyndns.org
  -- SIP/weboffice.dyndns.org-092ff310 is ringing
  -- SIP/weboffice.dyndns.org-092ff310 answered SIP/390641043101-092ee290
  -- Attempting native bridge of SIP/390641043101-092ee290 and SIP/weboffice.dyndns.org-092ff310
  -- Started music on hold, class 'default', on channel 'SIP/390641043101-092ee290'
  Oct 2 14:23:47 NOTICE[27220]: res_musiconhold.c:533 monmp3thread: Request to schedule in the past?!?!
  Oct 2 14:23:47 NOTICE[27220]: res_musiconhold.c:533 monmp3thread: Request to schedule in the past?!?!
  Oct 2 14:23:51 NOTICE[27230]: chan_sip.c:7026 get_refer_info: Supervised transfer requested, but unable to find callid 'f1d4a57c-f11e291e-22ed860d@
  2.168.1.230'. Both legs must reside on Asterisk box to transfer at this time.
  -- Stopped music on hold on SIP/390641043101-092ee290
  == Spawn extension (from-unidata, 100, 1) exited non-zero on 'SIP/390641043101-092ee290'
  Oct 2 14:23:58 ERROR[27281]: cdr_csv.c:237 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or director
  Oct 2 14:23:58 ERROR[27281]: cdr_custom.c:129 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : No such file or d
  ectory
   
  Here is my sip.conf 
   
  [general]
srvlookup=yes ;allows DNS lookups of server names
register => 390641043101:password at sip.uni.it/100
allowexternalinvites=yes
disallow=all
allow=g729
allow=alaw
context=outbound
   
  [sip.uni.it]
type=peer
disallow=all
allow=g729
allow=alaw
dtmfmode=rfc2833
host=sip.uni.it
fromdomain=sip.uni.it
insecure=very
qualify=yes
fromuser=390641043101
authuser=390641043101
username=390641043101
secret=password
canreinvite=no
context=from-unidata
   
   
  Here is my extensions.conf
   
  [outbound]
exten => _0.,1,Dial(SIP/${EXTEN}@sip.uni.it,60,r)
exten => _X.,1,Dial(SIP/${EXTEN}@weboffice.dyndns.org,60,r)
   
  [from-unidata]
exten => 100,1,Dial(SIP/530 at weboffice.dyndns.org)
   
   
  Thanks to all and regards,
  Francesco.

       
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