[Home]

Summary:ASTERISK-15590: Park ed call slot annoucement is not heared
Reporter:Ronald Chan (loloski)Labels:
Date Opened:2010-02-07 14:52:57.000-0600Date Closed:2011-06-07 14:08:22
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Applications/app_parkandannounce
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Hi! Good day, today when i about to test asterisk SVN 1.6.2, parking lot slot announcement is never heard, but it is working as it should.

I'm not using a features.conf transfer here, the call was parked via a built in transfer button on eyebeam softphone.

If you need more information just feel free to ask...


Best regards,

Ronald

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

Topology:

Incoming Call

DAHDI -> * -> SIP -> PHONE

[Feb  8 12:44:23] NOTICE[32414]: chan_dahdi.c:8672 ss_thread: Got event 18 (Ring Begin)...
[Feb  8 12:44:24] NOTICE[32414]: chan_dahdi.c:8672 ss_thread: Got event 2 (Ring/Answered)...
   -- Executing [s@from-pstn:1] NoCDR("DAHDI/3-1", "") in new stack
   -- Executing [s@from-pstn:2] Answer("DAHDI/3-1", "") in new stack
   -- Executing [s@from-pstn:3] Ringing("DAHDI/3-1", "") in new stack
   -- Executing [s@from-pstn:4] Wait("DAHDI/3-1", "1") in new stack
   -- Executing [s@from-pstn:5] WaitExten("DAHDI/3-1", "2") in new stack
   -- Timeout on DAHDI/3-1, continuing...
   -- Executing [s@from-pstn:6] NoOp("DAHDI/3-1", "Incoming call from : ") in new stack
   -- Executing [s@from-pstn:7] Dial("DAHDI/3-1", "Sip/102,60,m(default)") in new stack
 == Using SIP RTP CoS mark 5
   -- Called 102
   -- Started music on hold, class 'default', on DAHDI/3-1
   -- SIP/102-0000000a is ringing
   -- SIP/102-0000000a answered DAHDI/3-1
   -- Stopped music on hold on DAHDI/3-1
   -- Started music on hold, class 'default', on DAHDI/3-1
 == Spawn extension (from-pstn, s, 7) exited non-zero on 'DAHDI/3-1'
   -- Started music on hold, class 'default', on DAHDI/3-1
 == Parked DAHDI/3-1 on 701 (lot default). Will timeout back to extension [from-pstn] s, 7 in 45 seconds
   -- Added extension '701' priority 1 to parkedcalls (0x8ec7d38)
   -- <SIP/102-0000000a> Playing 'digits/7.slin' (language 'en')
   -- <SIP/102-0000000a> Playing 'digits/0.slin' (language 'en')
   -- <SIP/102-0000000a> Playing 'digits/1.slin' (language 'en')
 == Using SIP RTP CoS mark 5
   -- Executing [701@from-internal:1] ParkedCall("SIP/102-0000000b", "701") in new stack
   -- Stopped music on hold on DAHDI/3-1
   -- Channel SIP/102-0000000b connected to parked call 701

Asterisk SVN-branch-1.6.2-r245193, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details.
=========================================================================
Connected to Asterisk SVN-branch-1.6.2-r245193 currently running on pbx (pid = 31517)
Verbosity is at least 3
pbx*CLI>

SIP definition
[basic-options](!)                ; a template
       dtmfmode=rfc2833
       context=from-internal
       type=friend
       qualify=yes

[natted-phone](!,basic-options)   ; another template inheriting basic-options
       nat=yes
       directmedia=no
       host=dynamic

[public-phone](!,basic-options)   ; another template inheriting basic-options
       nat=no
       directmedia=yes

[my-codecs](!)                    ; a template for my preferred codecs
       disallow=all
       allow=g729
       allow=ulaw

[ulaw-phone](!)                   ; and another one for ulaw-only
       disallow=all
       allow=ulaw

[100](natted-phone,my-codecs)
       defaultuser = 100
       secret = 100

[101](natted-phone,my-codecs)
       defaultuser = 101
       secret = 101

[102](natted-phone,my-codecs)
       defaultuser = 102
       secret = 102
Comments:By: Ronald Chan (loloski) 2010-02-07 15:02:29.000-0600

I don't think this is really necessary but i will post it here anyway a snippet dialplan


[from-internal]

exten => _1XX,1,Dial(Sip/${EXTEN},60)
exten => _1XX,n,Hangup

include => parkedcalls
include => outgoing
include => utilities
By: Ronald Chan (loloski) 2010-02-07 15:17:57.000-0600

When i scanned the bug tracker i see similar something similar post ASTERISK-13476 and according to terry w., if you used a SIP transfer built in XFER button in your soft phone/sip phone it is normal not to heard the announcement and that is the expected behavior?

If yes, then please close this bug... thanks


Best regards

Ronald
By: Leif Madsen (lmadsen) 2010-02-08 11:15:11.000-0600

Ya, I was thinking that the built in transfer is likely a blind transfer, in which case you wouldn't hear the announcement. I think that is expected behaviour here.