<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
I've set transfer = no for all channels in chan_dahdi.conf, but I still
have the same <br>
<br>
[channels]<br>
context=from-pstn<br>
signalling=fxs_ks<br>
rxwink=300 ; Atlas seems to use long (250ms) winks<br>
usecallerid=yes<br>
hidecallerid=no<br>
callwaiting=yes<br>
usecallingpres=yes<br>
callwaitingcallerid=yes<br>
threewaycalling=yes<br>
transfer=no<br>
canpark=yes<br>
cancallforward=yes<br>
callreturn=yes<br>
echocancel=yes<br>
echocancelwhenbridged=no<br>
;faxdetect=incoming<br>
;echotraining=800<br>
callgroup=1<br>
pickupgroup=1<br>
relaxdtmf=yes<br>
<br>
This is the log of the second call. I am pressing flash to make the
transfer, the bad thing is that a short on-hook situation simulate
that flash, and are making this unwanted transfer.<br>
<br>
<br>
[Sep 30 07:17:41] VERBOSE[3237] logger.c: -- Called g2/16<br>
[Sep 30 07:17:42] DEBUG[3237] chan_dahdi.c: Sent deferred digit string:
T16w<br>
[Sep 30 07:17:43] VERBOSE[3237] logger.c: -- DAHDI/9-1 answered
SIP/101-087c9288<br>
[Sep 30 07:17:49] VERBOSE[3054] logger.c: -- Stopped music on hold
on DAHDI/8-1<br>
[Sep 30 07:17:49] DEBUG[3054] chan_sip.c: SIP transfer: Succeeded to
masquerade channels.<br>
[Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: New owner for channel 8 is
DAHDI/8-1<br>
[Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: master: 8, slave: 9,
nothingok: 0<br>
[Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: Stopping tones on 8/0
talking to 9/0<br>
[Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: Stopping tones on 9/0
talking to 8/0<br>
[Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: Making 9 slave to master 8
at 0<br>
[Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: Added 20 to conference 9/8<br>
[Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: Added 19 to conference 9/9<br>
<br>
<br>
<br>
<br>
Kevin P. Fleming escribió:
<blockquote cite="mid:4AC1FC2B.9050707@digium.com" type="cite">
<pre wrap="">Maurizio Faccio adinet wrote:
</pre>
<blockquote type="cite">
<pre wrap="">I own a TDM2400 board, with three FXO modules and one FXS.
I'am having trouble with analog sip phones, from two different
equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202),
sometimes when I am calling someone, then I press flash, and then call
someone else, both calls stay connected after I hang up.
</pre>
</blockquote>
<pre wrap=""><!---->
That's because you have just completed a flash-hook based transfer of
the first call to the second call. If you don't want this feature, set
'transfer=no' for the relevant channels in chan_dahdi.conf.
</pre>
</blockquote>
</body>
</html>