similar to: asterisk-users Digest, Vol 64, Issue 52

The script runs fine command line. I have edited in the past to try as /usr/bin/php -q and it didn't help. Right now, it's not even reading the changes. I must be missing something very obvious... LN asterisk-users-request at lists.digium.com wrote: > Message: 16 > Date: Wed, 24 Jun 2009 17:17:59 -0400 > From: "Juan E. Rodr?guez" <jerdguez at gmail.com> >

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), I have several extensions that can register 2 separate devices (chan_sip) ( FreePBX calls this Devices & Users mode : Users are extension/internal number, devices are the 'SIP Accounts' for the internal 'endpoints' ) (this

Hello everybody I am trying to connect my asterisk to a payment equipment trough PSTN. I have a TDM400P card with an fxs module an the equipment use modem to send data! I was thinking to implement a software data modem in asterisk, but I found out that there is just faxmodem for asterisk, Is anyone here know something about software data modem working with asterisk to help out? Thanks,

adding gsm or just comment out the disallow and the 2 allows. (your = recipient is using a codec that isn't ulaw or alaw). =20 _____ =20 From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel = freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

expected context is valid (may not work on 1.2, I started this ride at 1.4 and therefore have no backward knowledge). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David Nickel Sent: Wednesday, May 05, 2010 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hash Dial

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

--- span_1 = DAHDI/g1 1,1,dial(${span_1}/${EXTEN:0}) --- I can only presume some form of precedence overrides the group configuration in the recent asterisk installs and not for the servers set up earlier. On 26/5/09 4:01 PM, "Kal Feher" <kalman.feher at melbourneit.com.au> wrote: > Ok I've solved the problem. I do not think it was as switchtype issue after > all as

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> >

Hi?Danny, Thank you for your prompt response. The way you are suggesting is great .?Infect?asterisk have its own functionality that if user presses *1 during meetme conferencing asterisk automatically unmute that user and user comes in talking mode.But it is not?fulfill my need. There is and issue that if 3-4 user presses *1 at the same time than how can i decide that who is asking the question

takes precedence over a queue's defined moh class. --=20 Thanks, --Warren Selby, dCAP http://www.selbytech.com --000e0ce0494051d402049b4247c1 Content-Type: text/html; charset=windows-1252 Content-Transfer-Encoding: quoted-printable <div class=3D"gmail_quote">On Tue, Feb 1, 2011 at 10:20 AM, Danny Nicholas = <span dir=3D"ltr">&lt;<a

Date: Fri, 1 Oct 2010 18:40:40 -0300 From: Rodrigo Lang <rodrigoferreiralang at gmail.com> Subject: Re: [asterisk-users] AMI Originate To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <AANLkTikV+32vKVSkAFmkDciOPn+rO=k3jYJmsZLNj1QS at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" 3

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

I have the very same situation in one of my networks. To solve this you can dial out from the softphone and to move call to the phone you can simply transfer call to the same user (just if you were transferring call to yourself and the other device will ring. While, as you notice, you cannot dial a device, you can surely call your user to tranfer from a device to another. Please note that call

Hi all, Thanks to the previous replies that helped me with this before, but I got side-tracked in the middle of trying to figure this out, so apologies for posting the same issue. I use a Nokia e71, with an asterisk server and am having an issue dialing certain numbers. When I attempt to dial a local number, like xxx-xxx-xxxx, I cannot connect. What shows in the asterisk debug is the