Displaying 20 results from an estimated 2000 matches similar to: "picking up a ringing extension"
2003 Jul 09
2
error on web page for msn
Hi everybody,
I'm trying to use msn with * and for that, I'm reading
all information on the mailing list. You used to
recommend the page http://mcleod.pbx.nq.net/msn/, but
I always get an error while opening. Has it changed?
Is there another one?
Thanks
cmayor
___________________________________________________
Yahoo! Messenger - Nueva versi?n GRATIS
Super Webcam, voz, caritas animadas, y
2003 Jul 09
1
more abou msn
Hi,
Talking about messenger,,, it's still necesary to do
HKEY_CURRENT_USER\Software\Microsoft\MessengerService\Corp2PC_Phone
equals to '1' ??? But it's still sending the '+'
digit, so it's necesary to stripMSD?
Thanks a lot
cmayor
___________________________________________________
Yahoo! Messenger - Nueva versi?n GRATIS
Super Webcam, voz, caritas animadas, y m?s...
2003 Jul 07
1
callgroup and pickupgroup
Hi,
I asked a time ago what were callgroup and pickup
group used for. I have done some proofs and all, and
I'm not sure if I have pick the idea up well!!
That's what I understand:
For example: group=1 callgroup =2 and pickupgroup=2
and my phone is a membership of the group 1.
that's mean that when a phone that belong to group 2
is ringing, I'll be able to answer this call dialing
2003 Jul 08
2
Transfert call
Hi,
A question about transfert.
How can I make transfert with the the person who call.
X call Z and X transfert Z to Y.
I only succeed to do X call Z and Z transfert to Y.
If someone have a solution it will be very good =)
regards
Rattana
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2007 Mar 21
7
polycom random reboots
Hi,
At one location we have a user whose Polycom IP430 suffers from erratic
reboots. We swapped his phone for a brand new one, but the problem
remains.
Has anyone seen that?
2009 Jan 21
4
integration with Microsoft CRM?
Hi,
How hard is it to integrate asterisk with Microsoft CRM?
Thanks for any suggestions, pointers, etc.
2007 Mar 18
3
how can I use rsync between 2 accounts?
Hi,
I have 2 linux accounts on different machines (same login, same password).
Can you please tell me how I use rsync directories between 2 accounts?
Thank you.
2003 Jul 08
1
chanh323 dialling
what is the format for an h323 entry in the dialplan?
can I use chan_h323 without compiling anything else
or should I compile oh323?
basically what's the best way :)
cheers
Dave
2003 Jul 11
3
What does "callerid=" in sip.conf do?
Hi
since "callerid=" in sip.conf doesn't set the Caller ID, I suppose it
must be there for some other reason.
Is this a not-yet-working feature for future releases of Asterisk?
If not, what does it actually do?
thanks
regards
bk
2003 Jul 08
2
oh323 problem (small one)
I have just compiled & installed the latest oh323, on a fresh asterisk
installation
however using a previously working oh323.conf file.
When I try to dial an outbound oh323 call I get the following error :
-- Going to extension s|1 because of immediate=yes
-- Executing Wait("Zap/1-1", "1") in new stack
-- Accepting call from '21382890' to 's'
2008 Nov 11
2
TE410P alarms stay RED with 1.4.22
Hi,
I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but
then my TE410P alarms stay RED and no zap channels can be created, even
if they are correctly listed by "zap show channels". I tried adding
"dahdichanname = no" to asterisk.conf's [options] to no effect.
Going back to 1.4.21.2 brings my alarms back to OK.
This is with zaptel
2009 Jul 24
2
how to match "no callerid" in 1.6 ?
Hi,
This used to work fine in 1.4:
exten => 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten => 2131/,n,Playback(no_unknow_callerid_here)
exten => 2131/,n,Hangup
And now, after upgrading to 1.6.1.x it matches every callerid.
Did something change?
Thanks,
2008 Dec 20
2
autolinking URL's
Hi,
Is there a way to have markdown automatically convert obvious (http,
mailto) URL's to links?
i.e: http://example.com -> <a href="http://example.com>http://example.com</a>
Thanks,
--
http://www.critikart.net
2005 Mar 28
3
can a sip.conf stanza be shared by several phones?
Hi,
If several phones register to the same sip.conf section what will happen
with a "Dial SIP/shared" in asterisk?
All phones ringing and the first one to answer gets the call?
Undefined behavior?
Thanks,
--
Jesus is coming! Everyone look busy!
2006 Nov 08
1
HANGUPCAUSE for unalocated number?
Hello,
On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an
unalocated number? I always get 3 (no route) which is less than helpful.
2010 Sep 09
2
is a "- *.ext" filter overriden by a later "+ *.ext"
Hi,
In our backup script we sometimes would like to override the common
(i.e: static) excludes filter list. For example we exclude "- *.ext" for
all backups but would like to include "+ *.ext" only for 'local'
backups.
Are such entries supposed to cancel each other? How can one override an
earlier exclude in a filter list?
Thanks,
2004 Mar 15
1
Megre ext3/ext2 partitions?
Hi!
Is it possible to merge two ext3/ext2 partitions into ONE ext3/ext2
partition?
--
Ralf Hildebrandt (Im Auftrag des Referat V a) Ralf.Hildebrandt at charite.de
Charite - Universit?tsmedizin Berlin Tel. +49 (0)30-450 570-155
Gemeinsame Einrichtung von FU- und HU-Berlin Fax. +49 (0)30-450 570-916
IT-Zentrum Standort Campus Mitte AIM. ralfpostfix
2007 Mar 30
1
bad case of buzzing
Hello,
We are at wit's end on this. One (and only one) of our five asterisk
installation is giving us real headaches. Buzzing and/or choppy sound
interfere with conversations. I recorded some conversations with
monitor() and no problem whatsoever appear in the recording, while the
local user was hearing the buzz and half my words.
This is a 1.2.16 installation with mISDN but mostly using
2003 Sep 10
1
running * on a VPN gateway
If like me you run * on a VPN (or multihomed) gateway and want to serve
remote SIP clients, make sure you have
bindaddr = 192.168.0.1 ; or whatever is your box's private IP
otherwise * might bind to its public IP and send it as return address in
the SIP call setup, which will (should) be rejected by your firewall.
To * experts: might this setting interfer with NATed SIP clients?
--
I
2005 Jun 20
1
oneTouchVoicemail issue with Polycom 1.5.2
Hi,
After upgrading to 1.5.2 I no longer can directly access to my voicemail
by pressing the "Message" button, I have to go through the
"urgent,new,old" report first. The oneTouchVoicemail parameter is set to
1 but not taken into account apparently.
Anyone noticed that problem?