bruce bruce wrote:
Hi Everyone,
I can connect the Jabra GN2124 + GN2100 (smart cord) to the Aastra 53i
receiver port and I get a tone. But when I connect it to the headset
port there is no tone. I am running firmware 2.4 and I can't seem to
find that DHSG, EHS or whatever the setting
On Wed, 2010-08-25 at 17:42 -0400, Dan Journo wrote:
Hello,
I've posted about this a few months back but I didn't understand the
answer properly and only just got round to sorting it out.
My question is, when I dial out to a few numbers at the same time, the
CDR lastdata for the
Hello,
we have an asterisk (1.4.21.2) with ZAP and mISDN channels, the mISDN are 2
incoming GSM Modems, each with 2 simcards.
No, the mISDN line one and two are fine, but when I get a call on line 3
something with the time is wrong. Timeconditions fall through to off-hours even
if the time of
Hello list,
I have defined a new MoH-class in musiconhold.conf :
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes
;
*[106002]
mode=files
directory=/var/lib/asterisk/moh/106002
random=yes*
In sip.conf I have this commented out :
;mohinterpret=default
;mohsuggest=default
Hello,
I planning to use a web interface to send sms through Asterisk server.
I am planning to use php code which will interact with Asterisk Manager
Interface(AMI) and use Sms() application to send sms.
I am not sure whether it is the write way to do this. Anybody have any
suggestions or tips,
On Thu, 2010-08-26 at 12:10 +0200, Jonas Kellens wrote:
Hello list,
I have defined a new MoH-class in musiconhold.conf :
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes
;
[106002]
mode=files
directory=/var/lib/asterisk/moh/106002
random=yes
In sip.conf I have this
I had a similar problem and as far as I know, the asterisk server doesn't
know which of those numbers has answered your call.
If anyone knows any different, I'd like to know as well!
Got it!
I created a context that contained this:-
[outgoing_context]
exten =
On Thu, 2010-08-26 at 06:57 -0400, Dan Journo wrote:
I had a similar problem and as far as I know, the asterisk server doesn't
know which of those numbers has answered your call.
If anyone knows any different, I'd like to know as well!
Got it!
I created a context that contained this:-
So one shows as answered and the other doesn't?
Correct.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Thanks. That is one thing I really HATE about AASTRA - them confusing the
user with providing different setting level on the WEB UI and the PHONE UI -
very stupid.
But thank you and it works just fine.
-Bruce
On Thu, Aug 26, 2010 at 4:09 AM, Gareth Blades
list-aster...@skycomuk.comwrote:
There will be a brief outage of servers hosting Asterisk services on Saturday,
August 28, 2010 between 10am and 11am for maintenance. These services include
the following sites:
* packages.asterisk.org
* svn.digium.com
* svn.asterisk.org
* svncommunity.digium.com
* issues.asterisk.org
Please note that the timezone is -0500 GMT (Central Daylight Time, CDT).
Thanks!
- The Asterisk Team
On 10-08-26 10:32 AM, Asterisk Development Team wrote:
There will be a brief outage of servers hosting Asterisk services on
Saturday,
August 28, 2010 between 10am and 11am for maintenance.
Hi Everyone,
There are a few things I like in OrderlyStats, specially some graph
presentations and the fact that if agent puts someone on HOLD or PAUSE it
shows fine.
1 -But I see a lot of similarities in pricing, descriptions, wording on both
sites. Were these same projects forked out? or is it
On Thursday 26 August 2010 03:41:11 Raimund Sacherer wrote:
No, the mISDN line one and two are fine, but when I get a call on line 3
something with the time is wrong. Timeconditions fall through to off-hours
even if the time of the call is clearly inside business hours, here a log
excerpt:
As a test we built Asterisk v1.6.2.11 on a new server. This version of Asterisk
exhibits the same behavior. From ngrep's perspective we see an ACK followed
immediately by a BYE message. The user hears the recording being played, begins
to leave a message and is disconnected about 10 seconds
Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but
it doesn't seem to be doing anything as the script is still exiting on a
hangup and not completing properly. I am using 1.4.35 and have tried
various combinations. Can anyone shed any light on this?
Regards
Lee
--
I have searched for some time but I have not found an asnwer on how to
fix the CDR when a call is transferred. The problem is that if someone
dials a cell phone and then transfers the call to another extensión the
CDR for the cell call stops and there is no way to track that the call
was
We have a box running 1.6.2.11 on CentOS 5 using the RPM's from the Digium
CentOS repository. We just left a 60 second voicemail on the system and had
the full audio as well in the inbox. Not sure how your SIP configuration ties
your SBC in, but native users created via users.conf and
Can you post the CLI output showing the hangup/script failure?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer
Sent: Thursday, August 26, 2010 11:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
I've been getting complaints lately that callers to my IVR are pressing a
digit once but the system is responding as if they pressed it twice (once
for each of two consecutive menus).
I'm using an AGI script and logging all DTMF entries - and to the script, at
least, it looks like the digit
On 8/26/2010 2:55 PM, M S wrote:
Hi,
I've been getting complaints lately that callers to my IVR are
pressing a digit once but the system is responding as if they pressed
it twice (once for each of two consecutive menus).
I'm using an AGI script and logging all DTMF entries - and to the
On Thu, 26 Aug 2010, Lee Archer wrote:
Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but
it doesn’t seem to be doing anything as the script is still exiting on a
hangup and not completing properly. I am using 1.4.35 and have tried
various combinations. Can anyone shed
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Subject: Re: [asterisk-users] Use of AGISIGHUP
On Thu, 26 Aug 2010, Lee Archer wrote:
Hi, I am setting AGISIGHUP=no before running a Perl script via AGI but
it doesn?t
On Thu, 26 Aug 2010, Lee Archer wrote:
I am setting AGISIGHUP=no before running a Perl script via AGI but it
doesn?t seem to be doing anything as the script is still exiting on a
hangup and not completing properly.
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
We've actually had issues with Flowroute in the past where DTMF was a
constant issue. My best suggestion for course of action is find another
provider. NexVortex is pretty solid all around. They also had the quickest
recourse for when GNAPS went bottoms up last month and sent pretty much all
VoIP
First off, let me first say that this is not a one-way audio problem.
Sometimes I can get 'her' to speak to me, other times I can't for a
long time.
I'm just using a very simple system to dump people into MeetMe().
Nothing fancy.
I do have the following in my modules.conf:
preload =
How were you able to determine that the far end was sending the digits in
RFC2833 plus SIP INFO?
On Thu, Aug 26, 2010 at 3:23 PM, Andres and...@telesip.net wrote:
I have seen this before. Upon careful analisys we saw that the far end
was sending the digits in RFC2833 plus SIP INFO (or
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