12 jan 2010 kl. 20.56 skrev David Gibbons:
snip
'w' is really only supported on channels where digit-by-digit dialing is
the norm, which generally means analog trunks (or digital trunks using
CAS signaling).
/snip
Thanks Kevin, that's what I figured (though not quite so concisely)...
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest version (1.6) . Can you please let me
know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk
1.6 ?
Please
12 jan 2010 kl. 19.47 skrev Danny Nicholas:
Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a
1/2 second delay before dialing, ww1234 a 1 second delay, etc.
Try it with 2 or 3 w's instead of 1...
I have no solution, but can only say this: a 'w' in a SIP dialstring
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest version (1.6) . Can you please let me
know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk
1.6 ?
Please
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest version (1.6) . Can you please let me
know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk
1.6 ?
Please
On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net wrote:
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest version (1.6) . Can you please let me
know what are the
My apologies for the multiple copies.
Had issues with a mailserver that somehow wasn't talking to DNS properly. Now
fixed. It behaved like Asterisk does sometimes, very poor when it can't connect
to DNS. Had power outage yesterday and I think that started it all...
Meanwhile, I tried to
On Wed, Jan 13, 2010 at 8:27 AM, Olle E. Johansson o...@edvina.net wrote:
My apologies for the multiple copies.
Had issues with a mailserver that somehow wasn't talking to DNS properly.
Now fixed. It behaved like Asterisk does sometimes, very poor when it can't
connect to DNS. Had power
13 jan 2010 kl. 09.26 skrev hadi motamedi:
On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net wrote:
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest
Le 12/01/2010 16:35, Tilghman Lesher a écrit :
On Tuesday 12 January 2010 04:44:36 Benoit wrote:
I just experienced another problem however i have two rnis cards, one
b410p and one te220,
while the later works prefectly i can't really make the first one work,
using DAHDI or mISDN
under
Yes it's actually quite simple to do. If you want, the free version of
QueueMetrics is able to do that from the Agent's page.
l.
2010/1/13 Zhang Shukun bit...@gmail.com
2010/1/12 Lenz Emilitri lenz.lo...@gmail.com:
You can list phones directly as static members of the queue.
i know i can
On Wed, Jan 13, 2010 at 8:49 AM, Olle E. Johansson o...@edvina.net wrote:
13 jan 2010 kl. 09.26 skrev hadi motamedi:
On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net
wrote:
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on
On Tue, Jan 12, 2010 at 05:53:02PM -0600, Carlos Chavez wrote:
On Tue, 2010-01-12 at 18:05 -0500, C F wrote:
Anyone on the list ever used it?
I'm trying to quote a system with 192 analog ports, one of the options
are the Xorcom 32 channel FXS USB Channel Banks.
Any input would be
I reply to your question below
1) I don't have a secret for that peer.
2) Obviously, the solution is to make the 'host' field static (in my scenario,
because the port is non-standard 5080, so no standard endpoint SIP can
register with that IPaddress:port) or specify a secret with
On Wed, 2010-01-13 at 10:42 +0800, Zhang Shukun wrote:
is there some function used to login a agent automaticlly like
agentlogin(agentname,agentpassword,phonenumber)?
Depends what version you are running.
AgentCallBackLogin() is deprecated and you should not use it.
But the feature can be
A little more information please... the PC501 has how many lines defined(the
phone has 3 definable, can be 1,2 or 3)? Calls are SIP or DAHDI or Mixed?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
On Wed, Jan 13, 2010 at 2:43 AM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Thanks for that. Looking at the RTP packets I can see two types as you
point out. The first appears to be the audio:
Real-Time Transport Protocol
10.. = Version: RFC 1889 Version (2)
Payload
On Wed, 2010-01-13 at 10:12 -0500, Kristian Kielhofner wrote:
On Wed, Jan 13, 2010 at 2:43 AM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
Thanks for that. Looking at the RTP packets I can see two types as you
point out. The first appears to be the audio:
Real-Time
On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
..snip..
I've not been able to get that out of them, but I don't *think* It's
Asterisk based because they say:
Unfortunately, our assistance with Asterisk is extremely limited. For
configuration
I found this on the myth-tv list.
Can we do the same thing with asterisk?
Cheers,
Dean
-Original Message-
From: mythtv-users-boun...@mythtv.org
[mailto:mythtv-users-boun...@mythtv.org] On Behalf Of Kenni Lund
Sent: Wednesday, January 13, 2010 11:44 AM
To: Discussion about
On Wed, 2010-01-13 at 11:13 -0500, Kristian Kielhofner wrote:
On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
..snip..
I've not been able to get that out of them, but I don't *think* It's
Asterisk based because they say:
Unfortunately, our
On Wed, Jan 13, 2010 at 12:07 PM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
That may contain sensitive data, such as SIP account/password details -
so I'll pass on that, but thanks for the offer.
Even if they are using auth it's challenge response and fairly
difficult to
Upgrade the phone. I ran into the same issue a year or so ago. There was
some setting that was screwed up in the config file and upgrading to the
newest version at the time fixed it. It was something like the call waiting
tone being 30 seconds of dead air.
-Original Message-
From:
By upgrade the phone I assume you mean upgrade the bios, not purchase a
newer phone?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Wednesday, January 13, 2010 11:25 AM
To: 'Asterisk Users Mailing
Yes. Newer bootrom and sip image and sip.cfg.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, January 13, 2010 12:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Hi All,
I made the decision recently to update some old call servers from
1.2.x to the latest 1.4.28.
I spent 2 weeks in the lab testing every production requirement for
these voice servers. Nothing too special, SIP in, database lookup,
SIP out, a couple of special applications, write CDR to
Hello List
I'm trying to figure out what is wrong between my asterisk and my NEC 2400
pbx
We have been trying to link them with a spare PA-24DTG
from the NEC, I'm able to call an extension on the Asterisk, however the
extension rings, and then immediatly hangs up
I traced it back to the debug of
SOLVED!
Correct me anybody if I'm wrong but I think SAS option is for WAN only not for
the case if AudioCodes MP and Asterisk are on the same network.
I was trying to configure the fail-over mode in scenarios:
- Asterisk sever goes down (doesn't happen very often, never happened to me but
it
hi ,all
when use the VoiceMail , all the directions all english. i want to
know is there some Chinese version of sounds available now?
or should i record it myself?
just like:
Here is what you can do with your mailbox using VoiceMailMain.
1 Old Messages
3 Advanced options
1 Send reply
2
Thanks for your reply.
I have read the Asterisk Realtime Architecture feature of Asterisk.
it says that we can save queue and queue_members in the database. and
queue_member don't need to login( because not support). and when
queue_member changed in database. don't need reload cant asterisk use
when use the VoiceMail , all the directions all english. i want to
know is there some Chinese version of sounds available now?
or should i record it myself?
http://www.voip-info.org/wiki/view/Asterisk+sound+files+international
Look under Chinese (Mandarin)
--
I am wanting to use the ISDN cause code on an Asterisk 1.6 server to
determine the status of a call attempt, where the call might not actually
connect. Reason is I am checking for valid telephone numbers from a list of
numbers, and I would like to know if the call has answered and cleared which
I
Thank you!
2010/1/14 Lee, John (Sydney) john@compuware.com:
when use the VoiceMail , all the directions all english. i want to
know is there some Chinese version of sounds available now?
or should i record it myself?
http://www.voip-info.org/wiki/view/Asterisk+sound+files+international
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