From memory - 'rtcachefriends=yes' should do the trick.
PaulH
On Wed, 2007-11-28 at 16:56 -0800, Daniel Hazelbaker wrote:
I am trying to get the presence/hints/BLF working along with Realtime
SIP but I never get any busy notification. core show hints always
shows the realtime sip user
Thanks... That was just what I needed.
But what about going the other way? How can I pass multiple values to a
function in func_odbc?
I can't use ARRAY as it can only be used to set variables, not read form them!
Doug.
- Original Message
From: Tilghman Lesher [EMAIL PROTECTED]
To:
hi all
my local PSTN use an different cid signalling format
[ring-DTMF-ring],and my asterisk server can not get the cid
info,allways say UNKNOWN.I try to use EX220 (caller id converter)
translate DTMF into FSK,but after that the signal looks like
[ring-DTMF-ring-FSK],asterisk still UNKNOWN .Is
What LAN and you using? ELAN or HSP Are you trying to connect to a signaling
server? Please provide Nortel config.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, November 28, 2007 2:06 PM
To:
I caught the wrong bus once, and ended up as part of a murder
investigation.
Let this be a lesson to everyone!
PaulH
On Wed, 2007-11-28 at 06:50 +, broadband Voice wrote:
Hi,
Can anyone assist me in resolving this problem? I installed the G729
on a 32 and just found out that the
duhhh !!
Patrick wrote:
On Wed, 2007-11-28 at 17:08 +0530, Benjamin Jacob wrote:
simultaneous calls??.. will this correctly ensure the last call
retrieved from such DB was indeed the last call received?
Look at the subject. He said *dialled* number, not received :)
Regards,
Patrick
Ron McCarthy wrote:
Asterisk 1.4 im guessing? I did not know the Snom's worked with that,
Ill have to check it out then!
Yes, Asterisk 1.4. Also, it has been a while since I've used this code with
something other than a Polycom phone, but it should be ok.
The way it is implemented in Asterisk
Agreed!
Polycom and Polycom and Polycom!!
On Aug 20, 2007 3:26 PM, Michael Graves [EMAIL PROTECTED] wrote:
Sorry to top-post..but I haft agree here. Polycom is the KING of this
sort of thing.
Also, there really is a difference beteen a desk phone and a
conference/borard room phone. Having
I'd rather use a PCI card to connect * to the POTS, and a hard-disk
instead of a CF card. Do you know of a similar, small form-factor
motherboard + case that would fit the bill?
Many of the thin clients fit the bill nicely. I've been using MaxSpeed
MaxTerm clients lately. Mainly because I
I am experiencing several entries in the queue_log
with a duration of NULL at the COMPLETEAGENT or
COMPLETECALLER event.
Any idea how this can happen?
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to
On Nov 28, 2007 4:48 PM, Anthony Messina [EMAIL PROTECTED] wrote:
you're not the only one
I added a few distros and modified the few entries to reflect the true choices.
The only problem now is survey fatigue has set in because I didn't
wait until it was ready. The initial poll got 100
I have installed an Asterisk 1.4 on Suse93 using a FritzCard.
Some calls are logged to the ISDN log, but Asterisk is not detecting
incoming calls.
I wonder whether some other device or process is preventing Asterisk
from gaining access to the isdn line?
Is there some way to ensure that only
[asterisk-users] Asterisk - Nortel Phone Switch
Date: Thu, 29 Nov 2007 07:52:17 + (GMT)
X-Mailer: sendEmail-1.52
MIME-Version: 1.0
Content-Type: multipart/mixed; boundary=MIME delimiter for
sendEmail-20854.4017086787
This is a multi-part message in MIME format. To properly display this
Still trying to make my Asterisk PBK talk to our Nortel Phone Switch (C15k).
Nortel did an upgrade which changed a bunch of things today, so I thought I'd
give it another shot. It looks like I'm much closer this time, but still no
go. Can't do calling in either direction. Anyone have any
I have a multi-homed setup, and haven't had any issues, though it's
two separate network segments.
What version of * are you using?
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled
Hi All;
I have an IP Trunk established between Asterisk and
the VoIP service provider, when call from my mobile to
the PBX and then enter the destination number to call
via the VoIP, I got a connection but the voice level
volume need to be increased, I am trying to find if
zaptel (diguim card)
101 - 116 of 116 matches
Mail list logo
