similar to: Problem with DTFM and complex international setup

Displaying 20 results from an estimated 2000 matches similar to: "Problem with DTFM and complex international setup"

2005 Feb 18
1
Timing device OpenBSD
Hi all, I've been searching the wiki and google for a couple of days now but cannot find any reference to a timing source on OpenBSD. I have * CVS-v1-0-02/15/05-21:54:52 (I always do a cvs -q up -Pd before compiling) running like a charm on OpenBSD 3.6 Now I want to setup some IAX trunks to work and 3 friends and some meetme rooms but it looks like I need a zaptel timing source. Anyone can
2005 Feb 22
4
does asterisk support menus?
Whenever some call comes in i want it to be automatically picked up and then it plays some message "Welcome to xyz, Press 1 for sales and 2 for support" and then it takes it to the particular extension of sales/support. can i achieve this thing using asterisk? Kindest Muhammad Muzzamil Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Feb 15
1
IAX2 bugs...
Has anyone had stability issues with IAX2. (Asterisk 1.0.5). reddwarf*CLI> iax2 show firmware Device Version Size iaxy 22 39344 I'm asking because in the last three weeks I've noticed the following two issues (on separate occasions): 1) Placed a phone call. Asterisk logs show the phone being answered and various files being Played back. But
2006 Mar 09
3
DTFM or FSK
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2005 Mar 23
2
*-1.0.7 DTFM => Not working
My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it works in version 1.0.5 (was working with 1.0.3). I'm using SPA-3000 and dtmfmode=inband -- #Joseph
2005 May 13
1
DTMF problems with International Calls
Folks, I've setup an IVR system that's been running fine (in production) for the last 6 months. It's primarily been used for North American calls. The problem I'm having is that for international calls, DTMF does not work. The tones just don't get detected. I tried a number of different codecs and options, but to no avail. I'm using IAX through my provider. The numbers
2005 Jan 18
2
Is an unregistered phone busy?
Asterisk seems to regard an unregistered phone to be busy. Is that correct? Is not an unregistered phone unavailable? It is odd to me that if someone dials an unregistered extension, then the dialplan jumps to busy and voicemail kicks in saying that the person is on the phone, when clearly they can't be if the phone hasn't registered. Any way around this?
2005 Mar 09
26
OT: Best DB
I know this is a bit off topic but we are using Asterisk :) Since this list is full of tech gurus w/ all different sorts of backgrounds, I thought I would get the best opinions here. We have several different switches and other telecom equipment at our facilities which all have their own proprietary cdr platforms, which are rather limited. The company I work for is looking to develop their
2005 Jan 24
1
zaphfc no callerid incoming to SIP phone butvisible in logfile
Try commenting out the line pritrustusercid = yes Or set it to 'no'. That worked for me. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jens Sent: Friday, January 21, 2005 7:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] zaphfc no callerid
2005 Feb 21
1
Canadian DIDs...
Anybody know a good IAX provider for Canadian DIDs? Mohit. -- Mohit Muthanna [mohit (at) muthanna (uhuh) com] "There are 10 types of people. Those who understand binary, and those who don't."
2004 Feb 03
2
IPKall->FWD->Asterisk
Hi Folks, I recently setup an asterisk system in order to provide a telephone phone system for my web hosting business at a very low expense. My problem is that DTMF tones are not being recognized when calling the IPKall phone number. Calling my server via FWD and IAXTel works out fine however. Has anybody experienced this with the IPKall service? are they not passing the DTMF tones or am I doing
2010 Jul 12
0
DTFM Detection issues
Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller
2004 Dec 06
1
Broadvoice - bad quality, dtfm mode
Hello, I am sorry that I post questios regarding Broadvoice here, but unfortunalelly their support is very very bad. The simply do not answer to any emails or telephones. Last week something happened to their system. I was not able to receive incommming calls etc. Now it is back, but the voice quality is terribe and the DTMF is not working.(Is the inbound mode the correct one?) Does anybody knows
2007 Aug 24
0
DTFM not recognise
Hello,Maybe I don't understand what DTMF in ASCII means but I can't make my record stop using this syntax in a PHP agi script :fwrite(STDOUT, "RECORD FILE /var/lib/asterisk/ENR/jeanpaul wav '#' 15000 BEEP s=3000\n");The php syntax isn't a problem because I really start recording, I have a beep, the record can't long more than 15sec and after 3sec of silence my
2005 Feb 16
4
DTMF inband detection improvement
Hi all, I have some probleem detecting DTMF send by a GSM phone, I'm using SIP with ulaw. do you know what are the options to improve the detection ? I'm using asterisk 1.05, is the CVS HEAD version had some improvement about DTMF detection? Florian.
2009 Aug 20
12
IPKall and FWD
We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/4206395a/attachment.htm
2004 Nov 22
1
Using IPKall and SIP with insecure=very
Hi all, I've got one of those cool free incoming IPKall phone numbers from www.ipkall.com. These numbers just connect to the SIP proxy of your choice, they default to Frreworld Dialup. You can use them with your own sip proxy on asterisk. My config for this is below. The trouble I'm having is the incoming calls do not seem to hit the section in sip.conf for the call. With sip
2009 Apr 06
2
IPkall
Does IPKALL still exist? I am after a free SIP trunk - who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -------------- next part --------------
2010 Oct 24
5
Integrating Asterisk 1.8 with Google Talk and Google Voice
Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks
2005 Jul 21
11
IAX over HTTP
For work environments where you only get HTTP or HTTPS access, what is the feasibility of doing IAX over HTTP? I have heard of projects such as stunnel. Has anyone tried something like this? I did a quick search but didn't come up with much.