similar to: Re: Match a Numer - then continue with, dialplan

> -----Original Message----- > From: Richard Lyman [mailto:pchammer@dynx.net] > Sent: Wednesday, December 20, 2006 4:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: David

> -----Original Message----- > From: David Gomillion [mailto:dgomillion@eyecarenow.com] > Sent: Wednesday, December 20, 2006 10:27 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > I think you're making it far too difficult. > > What I do is something like this: > > [outgoing]

asterisk-users-request@lists.digium.com wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >

OK, I've got my script all set up and running, but now Asterisk crashes when the digits are entered with the following error: Ouch ... error while writing audio data: : Broken pipe I just retrieved and compiled the latest CVS this morning, as well as the latest AGI perl module. Why won't the AGI->get_data() function work correctly? Joe Richard Lyman <pchammer@dynx.net>

Sorry that got accidentally sent incompleted, here's the full post: OK, here is the long drawn out description of how I am using Zap Barge and Monitor: Zapbarge(listen in on live calls): Very simple actually I just added this to my dial plan(extensions.conf): ; barge monitoring extension exten => 8159,1,ZapBarge exten => 8159,2,Hangup Then when you dial 8159 on

> -----Original Message----- > From: Tony Mountifield [mailto:tony@softins.clara.co.uk] > Sent: Wednesday, December 20, 2006 2:41 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > In article > <645FEC31A18FE54A8721500CDD55A7B6035D0C6C@mail.oneeighty.com>, > Douglas Garstang

I have just created a very rough (read hack-ish) version of a Polycom SIP phone configurator. It allows you to define phones, create registrations, and such. By describing stuff about users, I am attempting to divine what the configuration should be. This is a VERY early first step in that direction. Right now, it: 1. Parses the phone1.cfg file included with whatever version software you use.

I am looking for a large number (probably about 100 or so) low-cost phones that I can hang on the wall. I need the phones to use PoE. Do the Uniden phones support wall-mounting? These phones are not going to be high-usage; they simply need to be there in case of an emergency. Another question, along the same kind of lines, has anyone figured out how to keep the SoundPoint IP 600 receiver

I seriously doubt he'd know how to get on the 'Internets' -----Original Message----- From: Doug Crompton [mailto:doug@crompton.com] Sent: Wed 12/20/2006 8:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [asterisk-users] Re: Match a Numer - then continue with, dialplan On Wed, 20 Dec 2006, Michael Collins wrote: > After listing all of that,

I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the

I think you're making it far too difficult. What I do is something like this: [outgoing] include => internal include => longdistance ;Always include internal first, as matches from the first include ;will be used first. This allows you to make sure your internal ;extensions don't go out your trunks. [longdistance] ignorepat => 9; include => default; already included from

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

Hello, You can use ZapBarge as an extension in your dialplan to listen in on conversations going on in Zap channels(Zaptel device channels) As for recording you can use the Manager interface command StartMonitor to start recording of a Zap channel and StopMonitor to stop it. Zap channels are pretty much the only ones right now that you can directly monitor and record through Asterisk. If

I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them out? I have 5 systems to manage, and it seems that about the only common file is

I have spent some time digging through the archives for comments concerning Asterisk and monitoring systems, and I have found few results. check_asterisk.pl.gz (http://www.dynx.net/ASTERISK/misc-progs/) which gives an error on download, and has no further Google references astping.tar (http://www.dynx.net/ASTERISK/misc-progs/ and also in the mailing list archives) supposedly sends a query to

All, Many moons ago I set up Samba 3.x with LDAP (Fedora Directory Server) back end. Frankly, I forget what example configurations I worked off of, but the GIDs and SambaSIDs for well-known "Domain" groups (Admins, Computers, Guests, Users) start around 2512. Fast forward 6 months and we have a commercial NAS (EMC) in the domain. It has worked fine for basic home directories, but

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I''m using Mechanize to parse an extraordinarily malformed html page. After submitting a form like so: page = mech.submit(dform) The result I get back is truncated. I suspect that it''s because the source HTML looks like this: <html> <head> yadda yadda</head> <p>some text</p> <html> <table yadda yadda> My

I have just installed Rmpi on a Suse 9.1 linux cluster with openmpi-1.0.1. I am trying the example included below from the tutorial website. However, I keep getting the following error: > # Load the R MPI package if it is not already loaded. > if (!is.loaded("mpi_initialize")) { + library("Rmpi") + } > > # Spawn as many slaves as possible >