Displaying 6 results from an estimated 6 matches for "qwork".
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2003 Sep 16
8
Hangups after voicemail
Hi,
Try as I might, I can't get hangups detected on a Zap channel with loop start
lines. So, after someone leaves a voicemail and then hangs up, Asterisk
doesn't know it, exits VoicemailMain2, and loops back to the corporate
greeting, tying up the line even though the outside caller has hung up.
Therefore, I've added the following hideous hack - er, code - to voicemail2.c.
It
2006 Oct 30
1
new BackgroundRB
Hey Greg-
Yes I am sorry, the new architecture uses fork and named pipes and a
bunch of unix stuff to do its magick. Now you may be able to port it
to qwork on windows, but I don''t think it is possible :( I''m really
sorry about this but I need this thing to be as robust and solid as
it can be and in the end windows isn''t compatible. Now you may be
able to still work locally by installing cygwin and running the
backgro...
2008 Mar 24
0
Xen3.1 setup-problem: no networking in WinXP-domU
Hi,
I cannot get my network up in WinXP- and winVISTA-domU''s
I need some help, since I got lost in the google-jungle...
My system:
UBUNTU gutsy gibon-AMD64 on a intel core duo-system.
this is my /etc/xen/winxp.cfg:
#########
#
# Configuration file for the Xen instance debian.qwork.test, created
# by xen-tools 3.5 on Sun Mar 16 20:03:26 2008.
#
#
# Kernel + memory size
#
kernel = ''/usr/lib/xen-ioemu-3.1/boot/hvmloader''
builder = ''hvm''
memory = ''512''
device_model=''/usr/lib/xen-ioemu-3.1/bin/qemu-dm''
# D...
2003 Sep 11
0
Hangup Detection and BUSYDETECT_MARTIN
Hello,
I've got the following configuration:
2 X101Ps
Asterisk built with BUSYDETECT_MARTIN
busydetect=yes
busycount=10
callprogress=yes
signalling = fxs_ks
With this setup, the best I can do is get voicemail with 17 to 19 seconds of
silence tacked on at the end. Ideally, I'd like at most 2-5 seconds. Has
anyone had any success with this?
It seems that hangups are indeed detected,
2003 Dec 05
0
Native bridging with Polycom 600
Hi,
I cannot get two Polycom 600 phones to bridge natively. My sip.conf has
canreinvite=yes for both phones. They connect, and I can talk as usual, but
sniffing shows the RTP stream is routed through Asterisk.
The exact spot where the attempt to natively bridge fails is in rtp.c, line
1281 (CVS from October 8, 2003):
f = ast_read(who);
if (!f || ((f->frametype == AST_FRAME_DTMF)
2003 Dec 10
0
Native Bridging and Polycom 600 Solved
Hi,
The Polycom 600 phones do not natively bridge with Asterisk. I've solved the
problem, but I'm not sure how general it is, so I thought I'd ask this list
for advice.
It's necessary to use a recent Asterisk CVS for this, since there was a
problem with session versions in earlier CVS builds.
The problem now is the Via field. When the reinvite goes out, the branch
number