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Summary:ASTERISK-10211: Fix for #10599 breaks attended transfer
Reporter:Dmitry Andrianov (dimas)Labels:
Date Opened:2007-08-31 07:48:37Date Closed:2007-08-31 09:22:05
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Resources/res_features
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Commit 81369 breaks transfer (I tested attended one).
When I chechout 81367 everything works fine. As soon as I update to 81369 (only res/res_features.c changed), transfer stops working.

Background:
* 1011 SIP softphone
* 1002 IAX softphone
* Zap/g3 connected to legacy PBX

Scenaio:
1. 1011 calls 1002
2. 1002 answers
3. 1011 dials *2 (to activate attended transfer)
4. 1011 dials 7243. The dialplan routes 7xxx numbers to legacy PBX (Zap/g3) and dials last three digits there
5. Asterisk actually opens Zap and dials www243 there so the phone connected to 243 extension of legacy PBX starts ringing.
6. At the same time, 1011 hears "I'm sorry that is not a valid extension, please try again". If legacy phone actually answers to the ring, he hears just congestion/busy tone.


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

Below is console output for rev 81369 (atxfer does NOT work):

   -- Executing [1002@ael-default:1] Macro("SIP/1011-09b3a428", "stdexten|1002|SIP/1002&IAX2/1002") in new stack
   -- Executing [s@macro-stdexten:1] Set("SIP/1011-09b3a428", "ext=1002") in new stack
   -- Executing [s@macro-stdexten:2] Set("SIP/1011-09b3a428", "dev=SIP/1002&IAX2/1002") in new stack
   -- Executing [s@macro-stdexten:3] Set("SIP/1011-09b3a428", "options=tT") in new stack
   -- Executing [s@macro-stdexten:4] Dial("SIP/1011-09b3a428", "SIP/1002&IAX2/1002/1002|20|tT") in new stack
[Aug 31 16:59:25] WARNING[2029]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
   -- Called 1002/1002
   -- Call accepted by 192.168.10.202 (format gsm)
   -- Format for call is gsm
   -- IAX2/1002-1 is ringing
   -- IAX2/1002-1 answered SIP/1011-09b3a428
   -- Started music on hold, class 'default', on IAX2/1002-1
   -- <SIP/1011-09b3a428> Playing 'pbx-transfer' (language 'en')
   -- Executing [7243@ael-default:1] Dial("Local/7243@ael-default-9afd,2", "Zap/g3/wwww243") in new stack
   -- Called g3/wwww243
   -- Zap/22-1 is ringing
   -- Local/7243@ael-default-9afd,1 is ringing
   -- Zap/22-1 answered Local/7243@ael-default-9afd,2
   -- Local/7243@ael-default-9afd,1 stopped sounds
   -- Stopped music on hold on IAX2/1002-1
   -- Hungup 'Zap/22-1'
 == Spawn extension (ael-default, 7243, 1) exited non-zero on 'Local/7243@ael-default-9afd,2'
   -- <SIP/1011-09b3a428> Playing 'pbx-invalid' (language 'en')


For compaison, below is console output for rev 81369 (atxfer works):

[Aug 31 17:02:22] NOTICE[4077]: chan_iax2.c:7575 socket_process: Peer '1002' is now REACHABLE! Time: 7
   -- Executing [1002@ael-default:1] Macro("SIP/1011-08d4e620", "stdexten|1002|SIP/1002&IAX2/1002") in new stack
   -- Executing [s@macro-stdexten:1] Set("SIP/1011-08d4e620", "ext=1002") in new stack
   -- Executing [s@macro-stdexten:2] Set("SIP/1011-08d4e620", "dev=SIP/1002&IAX2/1002") in new stack
   -- Executing [s@macro-stdexten:3] Set("SIP/1011-08d4e620", "options=tT") in new stack
   -- Executing [s@macro-stdexten:4] Dial("SIP/1011-08d4e620", "SIP/1002&IAX2/1002/1002|20|tT") in new stack
[Aug 31 17:02:28] WARNING[4104]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
   -- Called 1002/1002
   -- Call accepted by 192.168.10.202 (format gsm)
   -- Format for call is gsm
   -- IAX2/1002-4 is ringing
   -- IAX2/1002-4 answered SIP/1011-08d4e620
   -- Started music on hold, class 'default', on IAX2/1002-4
   -- <SIP/1011-08d4e620> Playing 'pbx-transfer' (language 'en')
   -- Executing [7243@ael-default:1] Dial("Local/7243@ael-default-4f22,2", "Zap/g3/wwww243") in new stack
   -- Called g3/wwww243
   -- Zap/22-1 is ringing
   -- Local/7243@ael-default-4f22,1 is ringing
   -- Zap/22-1 answered Local/7243@ael-default-4f22,2
[Aug 31 17:02:44] NOTICE[4104]: res_features.c:1234 ast_feature_request_and_dial: Don't know what to do about control frame: -1
[Aug 31 17:02:44] WARNING[4104]: cdr.c:827 ast_cdr_init: CDR already initialized on 'Local/7243@ael-default-4f22,1'
Comments:By: Digium Subversion (svnbot) 2007-08-31 09:20:46

Repository: asterisk
Revision: 81403

------------------------------------------------------------------------
r81403 | file | 2007-08-31 09:20:44 -0500 (Fri, 31 Aug 2007) | 4 lines

(closes issue ASTERISK-10211)
Reported by: dimas
Don't pass through the stopped sounds frame.... just drop it.

------------------------------------------------------------------------
By: Digium Subversion (svnbot) 2007-08-31 09:22:05

Repository: asterisk
Revision: 81404

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r81404 | file | 2007-08-31 09:22:05 -0500 (Fri, 31 Aug 2007) | 12 lines

Merged revisions 81403 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r81403 | file | 2007-08-31 11:38:59 -0300 (Fri, 31 Aug 2007) | 4 lines

(closes issue ASTERISK-10211)
Reported by: dimas
Don't pass through the stopped sounds frame.... just drop it.

........

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