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Summary:ASTERISK-16011: Blind transfer problem
Reporter:Maciej Krajewski (jamicque)Labels:
Date Opened:2010-04-26 04:24:40Date Closed:2010-04-26 10:47:43
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Resources/res_features
Versions:Frequency of
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Description:I've noticed a problem with blind transfer through features.conf.
When A is calling to B and A transfers the call to C everything is ok.
However, if B make the transfer the call is disconnected after the 'pbx-transfer.gsm', also the 'pbx-transfer.gsm' is in English, and channel language variable was set to pl earlier (if A makes the transfer the 'pbx-transfer.gsm' is in Polish).

****** ADDITIONAL INFORMATION ******

== Using SIP RTP TOS bits 136
 == Using SIP RTP CoS mark 4
 == Using SIP VRTP TOS bits 136
 == Using SIP VRTP CoS mark 4
 == Using UDPTL TOS bits 136
 == Using UDPTL CoS mark 4
   -- Executing [12@CALLEX:1] GotoIf("SIP/test001-00000175", "0?3") in new stack
   -- Executing [12@CALLEX:2] Set("SIP/test001-00000175", "__ORGDEST=12") in new stack
   -- Executing [12@CALLEX:3] AGI("SIP/test001-00000175", "agi://127.0.0.1/call-processor") in new stack
   -- AGI Script Executing Application: (Set) Options: (CHANNEL(language)=pl)
   -- AGI Script Executing Application: (Set) Options: (CALLERID(all)=test001 test001<+48583509013>)
   -- AGI Script Executing Application: (Set) Options: (GROUP(in)=user2)
   -- AGI Script Executing Application: (Set) Options: (GROUP(out)=user3)
   -- AGI Script Executing Application: (Set) Options: (_NUMBER_A=48583509013)
   -- AGI Script Executing Application: (Set) Options: (_NUMBER_B=12)
   -- AGI Script Executing Application: (Dial) Options: (SIP/test002,45,tTwW)
 == Using SIP RTP TOS bits 136
 == Using SIP RTP CoS mark 4
 == Using SIP VRTP TOS bits 136
 == Using SIP VRTP CoS mark 4
 == Using UDPTL TOS bits 136
 == Using UDPTL CoS mark 4
   -- Called test002
   -- SIP/test002-00000176 is ringing
   -- SIP/test002-00000176 answered SIP/test001-00000175
   -- Started music on hold, class 'default', on SIP/test001-00000175
   -- <SIP/test002-00000176> Playing 'pbx-transfer.gsm' (language 'en')
   -- Stopped music on hold on SIP/test001-00000175
      > [INSERT INTO cdr ("calldate","clid","src","dst","dcontext","channel","dstchannel","lastapp","lastdata","duration","billsec","disposition","amaflags","accountcode","uniqueid","userfield") VALUES ('2010-04-26 11:14:32','"test001 test001" <+48583509013>','+48583509013','12','CALLEX','SIP/test001-00000175','SIP/test002-00000176','Dial','SIP/test002,45,tTwW',14,12,'ANSWERED',3,'test001','1272273272.380','A:+48583509013;SI:CALLEX/test001;CRG:1;DI:CALLEX/test002;B:-12;')]
   -- <SIP/test001-00000175>AGI Script agi://127.0.0.1/call-processor completed, returning 0
   -- Executing [h@CALLEX:1] NoOp("SIP/test001-00000175", ""po AGI ===================="h" "12"") in new stack
   -- Executing [h@CALLEX:2] Hangup("SIP/test001-00000175", "") in new stack
 == Spawn extension (CALLEX, h, 2) exited non-zero on 'SIP/test001-00000175'
      > [INSERT INTO cdr ("calldate","dst","dcontext","channel","lastapp","lastdata","duration","billsec","disposition","amaflags","uniqueid") VALUES ('2010-04-26 11:14:32','s','CALLEX','SIP/test002-00000176','AGI','HANGUP',14,12,'ANSWERED',3,'1272273272.381')]
Comments:By: Paul Belanger (pabelanger) 2010-04-26 08:03:22

Please provide debug logs (see below), sip.conf and features.conf too.

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
By: Maciej Krajewski (jamicque) 2010-04-26 08:38:34

Sorry, for lack of logs, I'll do it better next time.
I've cheeked once again, and the problem occurs in Asterisk 1.6.16 in 16.1.18 everything is ok. Ticket can be closed.
By: Paul Belanger (pabelanger) 2010-04-26 10:47:27

Closed per reporters request.