asterisk users - Oct 2006 - attended transfer unreliable (2nd try)

Is there really nobody who has any idea about this?
help would be really apreciated, as otherwise we're forced to buy a
conventional pbx

----------------
Date: 29.09.2006 15:33
Subject: attended transfer unreliable
To: asterisk-users@lists.digium.com

Hi,

running asterisk 1.2.9 with freepbx 2.1.1, I have a strange problem:
sometimes, call transfer works as expectet, and sometimes not. So far, I
couldn't figure out any pattern in this behaviour,

features.conf :
featuredigittimeout => 1500
atxfer => *3
-----------------------------
works:
# user enters *
Sep 29 14:52:14 DEBUG[21578] channel.c: Got DTMF on channel (SIP/230-a983)
Sep 29 14:52:14 DEBUG[21578] channel.c: Bridge stops bridging channels
SIP/210-859a and SIP/230-a983
Sep 29 14:52:14 DEBUG[21578] res_features.c: Feature interpret:
chan=SIP/210-859a, peer=SIP/230-a983, sense=2, features=18
Sep 29 14:52:14 DEBUG[21578] res_features.c: Set time limit to 1500
Sep 29 14:52:14 VERBOSE[21578] logger.c:     -- Attempting native bridge of
SIP/210-859a and SIP/230-a983
# user enters 3
Sep 29 14:52:15 DEBUG[21578] channel.c: Got DTMF on channel (SIP/230-a983)
Sep 29 14:52:15 DEBUG[21578] channel.c: Bridge stops bridging channels
SIP/210-859a and SIP/230-a983
Sep 29 14:52:15 DEBUG[21578] res_features.c: Feature interpret:
chan=SIP/210-859a, peer=SIP/230-a983, sense=2, features=18
# here is the transfer:
Sep 29 14:52:15 DEBUG[21578] res_features.c: Executing Attended Transfer
SIP/210-859a, SIP/230-a983 (sense=2) XXX
Sep 29 14:52:15 VERBOSE[21578] logger.c:     -- Started music on hold, class
'default', on SIP/210-859a
-----------------------------
doesn't work:
# user enters *
Sep 29 09:17:54 DEBUG[20534] channel.c: Got DTMF on channel (SIP/230-9e2a)
Sep 29 09:17:54 DEBUG[20534] channel.c: Bridge stops bridging channels
SIP/210-c701 and SIP/230-9e2a
Sep 29 09:17:54 DEBUG[20534] res_features.c: Feature interpret:
chan=SIP/210-c701, peer=SIP/230-9e2a, sense=2, features=18
Sep 29 09:17:54 DEBUG[20534] res_features.c: Set time limit to 1500
Sep 29 09:17:54 VERBOSE[20534] logger.c:     -- Attempting native bridge of
SIP/210-c701 and SIP/230-9e2a
#user enters 3
Sep 29 09:17:55 DEBUG[20534] channel.c: Got DTMF on channel (SIP/230-9e2a)
Sep 29 09:17:55 DEBUG[20534] channel.c: Bridge stops bridging channels
SIP/210-c701 and SIP/230-9e2a
Sep 29 09:17:55 DEBUG[20534] res_features.c: Feature interpret:
chan=SIP/210-c701, peer=SIP/230-9e2a, sense=2, features=18
# no transfer
Sep 29 09:17:55 DEBUG[17507] chan_sip.c: Stopping retransmission on
'b3RAnUNZRdrkYNrR@192.168.11.161'
of Request 102: Match Found
Sep 29 09:17:55 DEBUG[17507] chan_sip.c: Stopping retransmission on '
b3RAnUNZRdrkYNrR@192.168.11.161 ' of Request 103: Match Found
Sep 29 09:17:55 VERBOSE[20534] logger.c:     -- Attempting native bridge of
SIP/210-c701 and SIP/230-9e2a
----------------------------------
when we have a timeout, it looks different:
Sep 29 12:00:34 DEBUG[21122] channel.c: Got DTMF on channel (SIP/240-6746)
Sep 29 12:00:34 DEBUG[21122] channel.c: Bridge stops bridging channels
Zap/2-1 and SIP/240-6746
Sep 29 12:00:34 DEBUG[21122] res_features.c: Feature interpret:
chan=Zap/2-1, peer=SIP/240-6746, sense=2, features=18
Sep 29 12:00:34 DEBUG[21122] res_features.c: Set time limit to 1500
Sep 29 12:00:36 DEBUG[21122] channel.c: Got DTMF on channel (SIP/240-6746)
Sep 29 12:00:36 DEBUG[21122] channel.c: Bridge stops bridging channels
Zap/2-1 and SIP/240-6746
Sep 29 12:00:36 DEBUG[21122] res_features.c: Timed out for feature!

hope your can help me

Stefan
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Stefan Friedrich wrote:
> Is there really nobody who has any idea about this?
> help would be really apreciated, as otherwise we're forced to buy a 
> conventional pbx
Have you tried upgrading to 1.2.12.1 or 1.2 branch from SVN?  There have 
been a few fixes in the branch that may help.

You can get instructions towards the center of the page at:

http://www.asterisk.org/download

Doug


-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."

Doug Lytle wrote:
> Have you tried upgrading to 1.2.12.1 or 1.2 branch from SVN?
Transfer (rather, dynamic features in general) is broken in 1.2.12.1:
http://bugs.digium.com/view.php?id=7982
So you should try the version from the SVN branch.

Yours, Florian.