similar to: Whats the '411' on echo cancellation?

Is it possible to allow distinctive rings work for FXS ports as well? I need a certain FXS extension to ring a distinctive double ring. I modified zapata.conf appropriately for dring1,dring2 and it just Seems to ignore my updates. Do distinctive rings only work for FXO ports? Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Seniuk.vcf

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk

Hello all, I am trying to setup a local asterisk server with SIP & ZAP extentions with And IAX2 'switch' to another Asterisk gateway with a PRI. I have managed to Get it working correctly for calls coming in and out. However, CallerID only Seems to work when I dial out. However, everytime I receive a call, it shows Up as 'IAX Guest User' on a SIP client. How can I

Hello all, I have a TE410P hooked up to a single PRI. Incomming CallerID is fine, and Outgoing works as well. However, if I change my dialplan for an extension To do a 'follow you, follow me' or setup an auto attendant that rings extensions Thru to cell phones, the CallerID always shows up on the call reciever as '708'. When I do a verbose dump at the console, it appears that

Hello all, I updated from CVS a few days ago and noticed that my calls just cut out without reason. The CLI says this: -- Hungup 'Zap/3-1' It occurs without error or warning. Is their a bug in CVS asterisk or libpri? This never occurred before. Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Seniuk.vcf Type:

Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none

Message: 5 To: asterisk-users@lists.digium.com From: Doug Meredith <doug.meredith@skyridge.com> Date: Wed, 21 Jan 2004 20:05:19 -0400 Organization: Skyridge Systems Inc. Subject: [Asterisk-Users] Re: What technology could my phone company be using? Reply-To: asterisk-users@lists.digium.com >>Mark Hazlewood <lists@idontknow.com> wrote: >>Sounds like Centrex services, we

Hello all, I am working on setting up a basic Asterisk system (using Digium FXS and FXO cards, Polycom IP phones). I've got a few questions in regards to voicemail and was hoping that someone could give me some ideas... Right now we have three incoming POTS lines. There are times when all three lines are used. When the lines busy out, the incoming call is sent to a voicemail

The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies

anyone know of a solution where mobile handsets out roaming the pstn cellular network can be used and treated as full fleged centrex extentions, i.e. I can transfer a call that comes in on a wired centrex copper pair out to a cell phone and the cell phone can transfer the call back or vice versa where the cell phone recieves the call directly and can transfer to the office all without hairpinning

I haven't dealt with Centrex for a long time, and one of my customers is being courted heavily by a Sprint salesperson. Am I not correct in assuming that each "line" of Centrex corresponds to an "extension" in the PBX world? This site has 2 POTS lines and 5 extensions, and they told me that for the same thing they're paying right now (~$40/POTS line) they will be

Greetings, List. I have my Asterisk box setup with 8 Centrex lines that were "left over" from our old PBX system. My boss is asking me to set up Asterisk so that he can flash hook and make an outgoing call on the same line to have a 3 way call. This is what he wants to do: 1) Incoming call on his Centrex line 2) Flash hook and dial a new number (goes out the same line) 3) Flash

Hello, I had a partner of mine present a Centrex 21 brochure and ask how many of those features can I fulfill. There is nothing out of the ordinary, it's stuff like call hold, call forward, 3-way calling, etc. Has anyone assembled a how-to that shows how to configure PBX or Centrex type functionality? I found one in the voip-info wiki but only a couple of topics were filled out. Regards,

Hello I saw Ver1.2 is out. Whats new? Thanks for the hard work, David

I am attempting installation of spandsp on to an Asterisk installation on Linux RH9 the distribution i am using is that are URL http://ftp2.tootai.net - the README for which i have followed verbatim - my only issue on this was the target for the port.h / tif_dir.h / tiffiop.h files in the 'headers' folder of the distribtion i put these in the /usr/include folder based simply on the

For whatever reason I decided to implement the validator class tonight. It''s my first go at implementing the class. My first run at it looked pretty good. I just added a generic .i for it and it all compiled. Sadly, the samples didn''t work because the base Validator class can''t actually be used as a validator! You have to derive from GenericValidator. I

I wrote earlier: > Please tell me the obvious mistake I'm making here.... The problem was a lack of sleep. Sorry to have troubled the list. -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca

Hi, I'm looking at setting up an Asterisk PBX in our office, which gets its phone lines (digital signaling, analog voice) from the main campus, which uses Centrex. Does anyone know if this falls under analog or digital for hardware buying? I was looking at getting a Digium TDM-series, but apparently our lines aren't pots (due to the digital signaling). Could someone enlighten me a bit?