similar to: [Fwd: Re: IAX config documentation]

Is there any documentation on configuring IAX between * machines? I've noticed references to many topics in the config files, including: - dialplans - trunking - authentication - transfers But before I go and try to grok 8000 lines of source (in one file, no less) I was hoping that somewhere there exists even something like a man page that describes the configuration options.

I am looking to install a web interface for Asterisk to transfer calls and look who's on the phone. If anybody has a working web interface please let me know. I installed the www.asternic.com (operator) But when I bring up my web browser it says transferring data and does not bring a browser. -----Original Message----- From: asterisk-users-admin@lists.digium.com

How do you setup the timing in Meetme conference? I have a x100p and tdm4x card. When I dialing to my conference I get a request to schedule in the past error message. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 10, 2004 10:48 AM To:

wondering if anybody knows this......does shady dial work only with a zap interface or can it be configured to be used with SIP or IAX. Nauman -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, July 08, 2004 5:48 PM To: asterisk-users@lists.digium.com Subject:

I'm having a problem with IAX running through an intermediate asterisk box. Perhaps a small diagram will explain the situation better: *A ------- [cloud (public internet)] ------- *B --------[cloud (private network)]----------- *C Asterisk server's A, B, and C, are all connected together with IAX All asterisk servers are 1.6.0.6 Server A and B are geographically close, but connected over

Hey.. Your saying I can not use flash with ISDN ? What options to I have to transfer a call directly ? ( So I have a free line afterwords) >> What interface are you using? ZapBRI? if so you might be able to do the >> hairpinning if it is supported. Im not using any interface.. But if you know how to do that, let me know and I install that interface. Thx for your answer :) Gr?sse

swar sir, can u please unsubscribe me for your list b.regards jihad chalhoub --- 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

So, Tilghman has put a particularly useful patch in the bugtracker: streaming music-on-hold is now supported. You can now specify .mp3 streams to be played back as MOH in the various places where MOH is used. Hopefully, Mark will install into the main CVS tree shortly. http://bugs.digium.com/bug_view_page.php?bug_id=0000413 This allows you to use the very sophisticated mp3 streaming audio

Extreme N00b, I am getting the error message "a target does not exist" when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Forwarded back to the list so others might get the benefit of the answers, and I get fact checked by others. -------- Forwarded Message -------- > From: Lee <leeb00@gmail.com> > Reply-To: Lee <leeb00@gmail.com> > To: Steven Critchfield <critch@basesys.com> > Subject: Re: [Asterisk-Users] udev or not? > Date: Fri, 10 Dec 2004 13:00:29 -0800 > On Fri, 10 Dec 2004

Just change the install prefix directory in asterisk makefile to something like /usr/local/asterisk, and then you will see all the files needed for asterisk to run. > -----Original Message----- > From: Steven Critchfield [mailto:critch at basesys.com] > Sent: Saturday, 1 March 2003 22:03 > To: asterisk-users at lists.digium.com > Subject: Re: [Asterisk-Users] How could I install

I have two 1-700 numbers from IAXTel. Both get registered from the same Asterisk server. I can make and receive calls on each without any difficulty. What I can't figure out how to do is route the incoming calls differently based on which 1-700 number is dialed. I must be missing something obvious. Thanks -brian -------------- next part -------------- An HTML attachment was scrubbed...

Okay, so like this? PHONE1+AD0-SIP/2000 PHONE2+AD0-SIP/3000 PHONE3+AD0-SIP/4000 ALL+AD0AJAB7-PHONE1+AH0AJgAkAHs-PHONE2+AH0AJgAkAHs-PHONE3+AH0- Then you would have Exten +AD0APg- s,1,Dial(+ACQAew-ALL+AH0-,20) Is that right? I have read about the Macros but don't understand their use. Could someone provide an example? Sorry about the newby questions... This will hopefully be my

Hi, this is my 1.st response to this list, i hope this will work. I tend to agree with Steven since just allowing other termination digits probaly wont solve your upcoming the issues anyway. I use a wrapper around the 'get digit' which allows me to specify that the * digit repeats the menu but maxium 3 times and if the * star digit is used twice in sequence (without other digits inbetween

PoE, or 802.3af, uses a device detection routine to determine if the connected device needs power. The process, in greatly simplified terms, is as follows: 1. Detect link state 2. Send a pulse of a known frequency and intensity over the TX/RX pairs 3. Listen for reflection. 3a. No reflection- provide power 3b. Reflection- no power Devices that comply with 802.3af have filters designed

After downloading the latest CVS head and testing it with the Cisco 7960 (SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid audio dropouts. I'm quite sure my gateway provider is running an older version of Asterisk, and I suppose that this may be the root cause. But I mention the issue here because it seems like it would be a mistake to ship Asterisk 1.0 if it

Speaking as former Dialogic/Bayonne user who was frustrated for months with Dialogic's complexity and months of initial testing with GlobalCall only to use their many-years-old and very complex base Dialogic drivers(and eventually scrapping it all for Digium/Asterisk and being up one week later), I can tell you that Digium/Asterisk is definately the way to go. It is 1000 times easier to

Also lookup AGI The WiKi and via google by using this: site:lists.digium.com <some words> W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steven Critchfield Sent: Friday, March 04, 2005 2:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Placing a call

Today I got my first PRI installed. It literally took less than 5 minutes and the circuit was up and we were making calls. The T100P is performing excellent. The Linux/Asterisk box is running well and the quality is great. The line is from MCI and they did a great job. I know this is not the usual banter but I just thought I would share a good experience and throw out some props to Digium

I must admit, reading this: http://www.xiph.org/ogg/vorbis/openletter.html brought out the loudest laugh in quite some time. My kids, hearing the hub-bub, gathered around the PC to see what I was laughing at. I *attempted* to explain the new mp3 licensing terms to my youngest daughter. I finally hit pay-dirt when I explained to my kids the bait-and-switch, what Thompson Media is doing now,