similar to: Getting started with sample dial plans

Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds;

http://www.asteriskdocs.org/modules/tinycontent/index.php?id=4 Can I make a suggestion that some documentation is provided for the Tortoise CVS download of the asterisk docs. I've tried every combination and I cant get it to work. I'm assuming it must work otherwise it wouldn't have been listed but for 60 seconds more work it would be a bigger benefit to the asterisk

Hi, is it possible, that my * identifies himself as ip-phone? I.e. Im using a grandstrem 100 phone and if I use * as proxy, the authentification string should be changed. Im not sure where looking for this. Hfh, Oliver

A while ago on the download page on www.asterisk.org, there was a stable branch for the asterisk source tree. It seems to have disappeared now, at least the instructions on that web page are gone. What's the story on this? Can we have it back please? thanks __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages!

Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees

Hello all, I want to setup Asterisk to forward a call if the dialed extension is busy. I do not want to wait on the line until the extension timeout expired. What I want is when I dial am extension currently Busy (Talking with someone), asterisk inmediately forwards my call to an extension I previosly defined. Someone could help me? Any clue will be appreciated. Regards from Spain. Ismael

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Adrien Laurent > Sent: 02 August 2005 14:56 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] How to create a secret code to use > myasterisk@home server's long distance plan from a public phone > >

Hi, I get fascinated when I dial someone and get an IVR play " for accounts department press 1, for sales, press 2 or hold the line for an operator" and then have MOH play before the line is picked up at the desired extesion. Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction. Thanks

Does anyone have a good list of links to the various presentations at Astricon, specifically one including a link to the performance analysis by those guys from Belgium? I would love to get a closer look at their graphs because it was impossible to read them, and I was pretty close to the front! -- Kristian Kielhofner

Are there DUNDi CLI commands for Asterisk 1.4? I have searched google and I only see the dundi commands in Asterisk 1.6, although I see reference to them in older version's of Asterisk such as Asterisk 1.4 here: http://www.asteriskguru.com/tutorials/cli_cmd_14.html. When I view the CLI commands through help I don't see any of the dundi commands and there are errors when I run a command

Hi, I got my TDM11B and am trying to get it to work on my PC. However, I'm having difficulty getting the wcfxs driver to load. I've Googled this problem, and while there are others who have ran into the same problem, none of the solutions work for me. I would very much appreciate it if you could suggest things I could try to resolve this problem. Details follow. The PC is somewhat

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 My organization has a LAN now , and there almost a computer in each office for each employee . And in such a situation , what the most economic way to deploy a PBX with asterisk ? Is there good tutorials for me to learn how to do ? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla -

Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Asterisk Gurus! I have a very simple question. I've just started playing around with Asterisk and BSD box. I also have grandstream ip phone and installed asterisk from ports. Now I'm on my very first steps to configure Asterisk. The question is: " How do I make Asterisk communicate with my Grandstream hard phone?" Thank you in

Hi all, i need to study the SIP protocol. can anybody tell me about any ebook which could halp me understand the sip protocol, architecture, and how to read and understand the sip signalling when i use "sip debug" in asterisk? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have been having a strange issue with my Asterisk 1.2.1 server. I have a TDM400 for the three POTS lines I have and I can receive calls without any problems. But sometimes (not everytime, but 70%) when I dial out of those lines it drops a number and of course gives me the telco error message. How do I know that it's dropping? Because I thought it was a line problem and the tech came out

Dear all I am going to install 2 port pri card on asterisk but i dont know how to incomming call goes in to IVR and how to route call outside base on pattern match means if some one call on mobile phone then use PRI 1 and if call on landline phon call route through pri 2 how to make dission base on pattern number Rgds satish patel

I am trying to get to the bottom of audio clicks, pops, dropouts with my Asterisk server. These occur even when the system is under minimal load (e.g. 1 Zap device in a queue being played music on hold) and occurs with both Zap and Sip devices so isn't network related. The audio problems occur at the same time on all channels and seems to be when Asterisk "gets busy" and uses

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