similar to: FAX over IAX

Has anyone gotten the voicetronix boards to work with Asterisk, what would it take? Or does anyone know where I can get 4 ports or more fxs PCI cards that do work with asterisk? Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

Like most others on this list I also have some really annoying echo whenever a call goes out to the PSTN from a SIP phone... SNOM/Budgettone -> Asterisk -> X100P -> PSTN I have tried every echo canceler in the makefile and turned on and off aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and I can get it reduced to only a few seconds on the intro of the call and

Hello, Does anyone know what may be causing this? Asterisk was built from cvs tonight. Ztcfg also says us is an invalid tone zone. Anyone got some information on what this is, why is it happening, and possibly some solutions? [root@anistonetech zapata]# asterisk -f -d DEBUG[1024]: File config.c, Line 653 (__ast_load): Parsing /etc/asterisk/asterisk.conf DEBUG[1024]: File config.c, Line 653

Everyone, Just thought I would drop a line telling everyone here I have the software RxFAX/TxFAX up and running without any real problems. I did have to..... RH 9.0 1) Install an audio devel rpm 1) install libtiff from source, and copy over a bunch of include files to /usr/local/include 2) build/install spandsp 3) move app_rxfax.c and app_txfax.c to apps/ dir in asterisk source tree. 4)

Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

Hi, I just got a T1 interface for a AT&T (became Lucent) System75 (uses same cards as Definity). I would take a crack at making a cable but can't determine the pinout for a cable and it is not apparent from the board. Asking you guys makes sense as one of you may have one of these systems. The cable has a amphenol male D50 connector on one end and probably a rj45 on the other. I also

I'm wondering if anyone knows of ISPs with service that has QoS features that would be good to use with VoIP stuff. Granted, the QoS would only be supported as long as you stayed within their network, but it might be better than nothing. --Eric

I have a simple problem with sialing a SIP device. I'm SURE it's a syntax problem, but I dunno what it might be. Here are the debug messages: == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>) -- Executing Goto("Zap/1-1", "2111|1") in new stack -- Goto (default,2111,1) -- Executing Dial("Zap/1-1",

Hello, I resolved my echo issue using grandstream/estara etc etc sip phones and wcfxo interfaces from digium. I swapped out my via kt400 based msi kt4vl motherboard for an asus p4pe? i845? based motherboard and the echo has completly gone away along with aggressive suppressor option in the makefile. I hope this helps others. Brian J. Schrock Anistone Technologies, LLC 6926 Avery Rd. Dublin, OH

I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal exten=> definitions? John This e-mail was scanned and found clean by Monroe-Woodbury CSD

Hi, I'm buying an interface for my Toshiba DKT424 to connect to *. Initially it is a trial to route internal calls over our wan. Think of it as a tie trunk between 2 pbxs. I expect to get PRI from the telco our layout will be like this: telco_T1_PRI to * to various PBX or TDM or IAX to * to other pbxs My question is, is there benefits to connecting with PRI over a standard T1 within my

I just called 6610 from Zap/1 exten => 6610,1,Dial,Zap/1&Zap/2 (Don't try this with the handset pressed up against your ear) ...is this the call waiting indicator? Is it a file, can it be turned down - and does it get used anywhere else? These are both fxs ports. In zapata.conf changinging to callwaiting=no stops this. The text implies that this affects fxo ports. John This e-mail

Hi, I'm looking for 4 line phones that will see some heavy usage (they need to be tough) for office secretary/receptionists. Any recommendations? One additional consideration (not particularly for my current needs) is regarding the page/intercom feature that can be mapped to one of the analog ports - does anyone know of a 4 line phone that will do this? Thanks, John This e-mail was

I have a machine that crashes every so often. I believe the following macro is responsible (gotoif,$[${ARG3}] in particular). The macro works as expected: if ARG3 is defined - hop over assignment. But my hunch is that it gradually chews up memory. ; This macro is puts voicemail in an alternate mailbox (if ARG3 defined - otherwise Mailbox matches extension). [macro-stdexten] exten =>

Hi, I am setting up a hylafax server. From what I've read so far, hylafax supports CID numbers and names but currently does not support DID. I assume I can do something like this... [40faxDIDs] exten => _87[5-8]X,1,SetVar(CALLERIDNAME=${EXTEN}) exten => _87[5-8]X,2,Dial(Zap/g${hylafaxMODEMGROUP}) ...and use the CIDName variable in hylafax to route the faxes to the appropriate

Here is something I wrote years ago. I expect you can adjust it for your needs # cat remove_blank_vmail #!/bin/bash # remove_blank_vmail takes arguments as voicemail boxes and removes messages with audio files shorter then MINSIZE (in bytes) #---------------------------------------------------------------------- # Description: # Author: John Harragin Monroe-Woodbury CSD # Created at: Thu Nov 6

Hi, Can I make e&m wink start lines just wait for digits - instead of going to default? Someone else cleared a similar problem (as described below) on an fxo port with "usecallerid => no" but it is not doing the trick for me. In this case the line when straight to default which would be ok also. John I posted the stuff below about a week ago... I set up a t1 from my sys75 to

I would like to access VoiceMailMain2 skipping extension and password prompting if calling from a resource that has a mailbox defined. What variables can I use to retrieve the calling channel & calling extension (if it exists)? Here is what I'm trying to accomplish (of course ${CallingResourse.MailBox} is not a real way to retrieve this info)... exten =>

Hi all, I need to be able to delete a voicemail message using a program. Is is sufficient to simply delete the .wav and .txt files in the spool directory? Or do I need to also renumber the remaining files? For example, let say a given mailbox has 20 messages in it and I want to delete message number 5. Can I just delete the 2 files and expect that asterisk will renumber them? Or do I

Hello, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.