Displaying 20 results from an estimated 1000 matches similar to: "configure the sound for inbound calls"
2010 Mar 24
6
Restarting Asterisk using a script - Thanks to all -
Hi All,
I do have asterisk installed for a call center and I would like to know if
it is possible to create a scipt and execute it from a PC connected to the
Network without accessing the server. This script should restart asterisk
and another service related to aheeva.
The problem now is that each time I have to access using PUTY to the server
to start and run services manually.
Service
2010 Oct 22
3
Licensing of Default MOH
Hi,
I wonder if I may freely use the default soundfiles that came with asterisk
(fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server?
Are there any official sources of royalty free music?
--
Mvh,
Aurimas Skirgaila
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2012 Feb 09
4
checking if a phone number is UP
hi,
We have a phone number from third party provider which is used for inbound
calls. How could I monitor if this *phone number* is reachable?
the initial idea doesn't sound elegant:
- on my SIP server I set couple seconds of ringing before Answer().
- the monitoring server calls to that phone number for few seconds, checks
if it "hears" the ringing and hangs up the call.
**
I use
2011 May 16
1
AMD tweaking
Hi,
long time ago, I came up with an optimal configuration set for
my environment - good detection and little false positives. Unfortunately
some people are always being detected as Answering Machines.
I'm not up to re-adjust my precious balance of initial_silence/max_words/...
, so I'm thinking about to check if the pickup time is equal to the pickup
time when the same phone number was
2006 Apr 04
2
Any Aheeva Users?
Just looking for unsolicited thoughts on the Aheeva product? Anyone
have anything to say?
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
2010 Mar 29
1
is it possible to connect Digium TE420 and Cisco card?
Hello,
I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to
providers Cisco 2800 with VWIC-1MFT-E1 card.
the same card runs fine with another E1 provider.
TE420 led's lite green.
Message type: RELEASE COMPLETE (90)
< [08 02 80 ac]
< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
Location: User (0)
< Ext: 1 Cause:
2012 Feb 02
1
amd detect answering machine
Hi,
I have IVR and when I press 1, asterisk calls my mobile phone.
If my mobile phone is offline, asterisk transfers to asterisk voicemail.
I'd like asterisk detects my mobile voicemail and if my mobile voicemail answers, asterisk transfers to asterisk voicemail.
For that, I used AMD.
So I have problems ! Asterisk detects answering machine everytime!
How do I do please ?
extensions.conf
2011 Apr 20
2
issue with installtion asterisk
hello all,
I have installed centos 5.5 ( linux text) and I have updated it with
# yum install bison bison-devel================?ok
# yum install ncurses ncurses-devel==========?ok
# yum install zlib zlib-devel===============?ok
# yum install openssl openssl-deve=======?ok
# yum install gnutls-devel============ ==?ok
# yum install gcc gcc-c++============?ok
# yum install newt
2005 Sep 30
2
Echo Cancellation not working in Zapata.conf
I have echocancel=yes in zapata.conf but when I do a zap show channel 1,
I notice echo cancellation is turned off.
I followed the article that talks about the order in which the
statements need to be in zapata.conf to get echo canceling to work:
http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html
But it is still not working. Does anyone know how to get echo
2006 Mar 21
2
Problem with chan_iax.c implimentationcausesbadaudio?
All switches and routers give highest priority to traffic on IAX2 port
4569. We use DSCB values over the IP-VPN to prioritize it as well.
This did not change with the upgrade, as we can still see proper packet
coding.
The softphone is provided by our vendor Aheeva. It is the same IAX2
softphone they use in their own call centers. Funny thing is that they
say that moving to Asterisk 1.2.4
2009 May 29
1
IAX2 trunking with Older Asterisk version ?
Hi All,
Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and
asterisk 1.2.14 ?
i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but
it gave an error -
1.2.14 End - Error Msg
WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by
147.120.203.71: No authority found
1.2 END , IAX.conf
[trunk14]
type=friend
host=147.120.203.71
secret=test123
2010 Aug 04
1
Tweaking AMD in Asterisk
Hello ,
I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My
current values are
AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx
25-30 % of all answering machines.
Anybody have any suggestion to improve the accuracy of AMD.
Thanks
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2006 Mar 21
1
Problem with chan_iax.cimplimentationcausesbadaudio?
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, March 21, 2006 11:36 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Problem with
chan_iax.cimplimentationcausesbadaudio?
On Tuesday 21 March 2006 11:19, Adam Robins wrote:
> All switches and routers give
2010 Apr 06
1
SIP Dialplan Failover Solution
Hello list,
I need a hand to find the best dialplan failover solution when using two SIP Trunks.
My reasons to do failover are:
a) one of the two providers could be unreachable
b) both providers may be UP but one of them could return a SIP error message (maybe caused by DOWN E1s)
Googling I found a few possible solutions:
1.
2004 Nov 23
0
Zombie channels dropping lines
Hi all,
We are running Asterisk 1.0.0 with a TE410P. Very often we exerience
calls dropping in the middle of the call. I enable the full logging and
saw a couple of suspicious messages right before the hangup. Thos could
happen on a Zap-IAX2 bridge as well as on a Zap-Agent bridge... I see
Nov 23 09:08:36 DEBUG[-1274020944]: Bridge stops because we're zombie or
need a soft hangup:
2008 Oct 27
1
autodialed call forwarding via meetme or queue (was predictive dialer)
Also posting this question to people working on manager interface and
dialers.
I have a simple auto dialing script (using Originate) that forwards all
incoming calls to a queue full of waiting agents instead of a meetme
conference room. I use queues rather than meetme so I can leave the
automatic call distribution to the queue itself.
The problem is when the calls reach the agents, some of the
2005 Jun 07
2
PRI Lines not being answered (No User Responding)
Hello! Continuing my PRI saga - I have a PRI setup and appears to be
answering calls OK, but my carrier is cutting all the calls after 15
seconds. For example, when I call from my cell phone, it goes
straight to a busy signal - however the CLI shows the call coming in
and being answered. Additionally, when I call from another ground
line, it will ring once or twice, again show as answered, but
2015 Feb 26
1
issue with inbound route
hello liste
i have creat i trunk sip and inboun route for inbound calls the issue whe i
use the DID in inboud route i have a error No DID or CID Match.
but when i leave this DID field blank i can route the call without any issue
how can ido in order to use DID in route inboud "i use elastix"
Executing [s at from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID
2012 Feb 24
43
[Bug 46557] New: nouveau: nv40 display corruption in framebuffer and X lockups unless nouveau.noaccel=1
https://bugs.freedesktop.org/show_bug.cgi?id=46557
Bug #: 46557
Summary: nouveau: nv40 display corruption in framebuffer and X
lockups unless nouveau.noaccel=1
Classification: Unclassified
Product: xorg
Version: git
Platform: x86-64 (AMD64)
OS/Version: Linux (All)
Status: NEW
Severity:
2013 Aug 14
3
[LLVMdev] BranchInst comparison
Your question isn't clear; please restate what specifically isn't working.
-Eli
On Wed, Aug 14, 2013 at 11:57 AM, Rasha Omar <rasha.sala7 at gmail.com> wrote:
> or like this
>
> %cmp4 = icmp eq i32 %rem, 0
>
> br i1 %cmp4, label %if.then5, label %if.else7
>
>
> On 14 August 2013 20:08, Rasha Omar <rasha.sala7 at gmail.com> wrote:
>
>> Hi