[Asterisk-Users] SIP flashhook transfer

Mickey Binder mickey at comflex.dk
Wed Apr 7 04:48:01 MST 2004


Hello * users

I try to get SIP flashhook transfer to work properly in my setup. The
problem is that when I flashhook and then dial another extension I get some
really garbled sound in the end I flashhook from. The remote can hear me
just fine, I have threewaycalling=yes and transfer=yes in my sip.conf. Here
is a complete overview of the sequence when I try to transfer the call:

1. Caller 1 dials some number with our PBX as destination 
2. Employee 1 picks up the call 
3. Employee 1 wants to transfer this call to Employee 2 at our company
(But Employee 1 wants to talk to Employee 2 first, "Supervised transfer") 
4. Employee 1 flashhook his phone and Caller 1 gets Music-On-Hold 
5. Employee 1 call Employee 2 and Employee 2 confirms that he can take the
call.
But now a problem arises: 
6. When Employee 1 hangs up in order to let Caller 1 and Employee 2 have a
conversation, "Music-On-Hold" start for 
Employee 2 also. In this state Caller 1 has "Music-On-Hold" forever or until
he hangs up too.

The 2 employees are in the same context in extension.conf, below is a snip
of my sip.conf file:


; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls
disallow = all
allow = ulaw
threewaycalling = yes
transfer = yes
tos = 184

[43300634]
type=friend
secret=
host=dynamic
dtmfmode=inband
defaultip=10.1.1.254
callerid=34
callgroup=1
pickupgroup=1
restrictcid=yes

[43300645]
type=friend
username=43300645
secret=
pickupgroup=1
callgroup=1
dtmfmode=inband
host=dynamic
defaultip=10.1.1.6
callerid=45

Regards
Mickey Binder





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