similar to: Re: Asterisk-Users Digest, Vol 20, Issue 226

The only advantage is when you travel. Last year I took my wifi sip phone to Astricon in Madrid and everything worked as expected. I am just packing it and heading for Paris... Wojtek -----Original Message----- From: Mojo with Horan & Company, LLC [mailto:mojo@horanappraisals.com] Sent: Tuesday, June 20, 2006 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Please, unsuscriber, this group. regars Nestor Castillo ----- Mensaje original ---- De: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> Para: asterisk-users at lists.digium.com Enviado: viernes, 20 de julio, 2007 11:00:04 Asunto: asterisk-users Digest, Vol 36, Issue 61 Send asterisk-users mailing list submissions to

i have ordered 500s from tritechcoa.com several times over the past 4 months. great service and delivery, and the prices are lowish, only problem is, they add a $20 handling fee per phone, on top of phone price, and shipping, making the lower price not as good -----Original Message----- From: Mojo with Horan & Company, LLC [mailto:mojo@horanappraisals.com] Sent: Friday, July 15, 2005 12:01

Hiya, hope I don't bore anybody with this. There are certainly a lot of monitor-y things out there and they just didn't fit my need, so maybe this will fit someone's besides mine. http://horanappraisals.com/asterisk/pbxmonitor/ contains two files. one is a php script called pbxmonitor, and one is a flat file of extensions to extension name mappings of internal users. It

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all

Hello, This seems to be a weird one. I'm at work now and will get some more-verbose logs later when I get home if nobody has any ideas about what's happening here. I've got a tdm card with 1 FXO and 1 FXS. Asterisk is in the 1.2.x line, so is zaptel. astlinux to be specific. I can get the versions at home later if it might help. It's running on a silent epia 5000 board

In futzing with my contact directory on my polycom 501s, I realized I could put 'username@another_phone_ipaddress' to call the other phone -- without even touching the asterisk server. I suspect my phone (and the remote phone) don't even have to be registered to an asterisk server, as the username@ part doesn't seem to have to match what the the remote phone is actually

> -----Urspr?ngliche Nachricht----- > Von: Mojo with Horan & Company, LLC [mailto:mojo at horanappraisals.com] > Gesendet: Dienstag, 25. M?rz 2008 23:23 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [asterisk-users] Asterisk parking hold and > transferdigittimeout > > It seems that the dialplan comes into play. If your parking >

Hi all, I have a small query regarding ringing tones on an iaxy2. I have a customer who uses an iaxy to breakout to pstn via our *. However the customer complains that he gets no ringing tone whislt making calls, i just visited the site and can confirm this. I also have another customer who is presently in canada with an iaxy calling thru our * , he doesnt have this issue. I presume that the

Hi.. I have to setup an extension in a remote location that will use a cordless analog telephone.. I am looking at the IAXY to do this for me..Basically the data path will be as follows... [Asterisk] == (NAT) == {Internet} == (NAT) == ATA -- Handset Since there are two NAT boxes in the path I know SIP won't work.. I also don't want to move the Asterisk box to the internet side of the

Hi, I have questions about the Polycom 601 and side card.... 1) In the side card the lights all time off... But all functions it's ok. I need help with extension module of polycom... All works fine... But lights not work.... So... I don't know when any person or extension is busy... Any ideas? , Olger On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com"

Mojo with Horan & Company, LLC wrote: > And it makes *clear* calls assuming you're within allowable range. > Speakerphone seems to work well too. I meant to mention that the DTMF tones and dialtone sound like they're played at such a high volume that they clip through the handset's speaker. DTMF is rfc2833, so what I'm hearing through the handset isn't affecting

On this page http://www.voip-info.org/wiki-Asterisk+record+calls there is "Example by Mojo". I have done everything he said and I have sox package installed. [root@pbx recordings]# sox -help sox: Version 12.17.7 ... When I open this web page http://10.0.0.26/recordings/index.php I get this: No Recordings Found And there are recordings in /var/spool/asterisk/monitor Do I have to do

Hello-- In the interest of Symmetry, does anyone else in the world see any need for a device like the IAXy (or the SIP ones from other manufacturers, like the ATA186), but one that presents an FXO interface instead, so it can be connected not to phones, but the PSTN? murf

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216

Hi. Does anyone know about a simple one-fxs ATA with pulse dialing support that can work with Asterisk? A SIP one would be ok. I've been told that the Digium S101i IAXy does support pulse dialing; although it's a iax2-only ata it could be enough. I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards

Hi, I have three Iaxy devices (s101i) parts. Two of them seem to work fine. The third plays a loud repeating click sound when an analog phone is plugged in. I can provision all of them, and make calls to all of them. The clicking one will blink when a call is incoming, but no audio from the call can be heard on the handset, and the caller only hears silence. The same handset works on the

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Hi all, getting a server going wiht a few TDM400's and some phones, and some IAXys too I haven't heard any issues about AU phones being able to RING in Australia, like the problem in the UK with ring capacitors on the BT system. Are there any problems like that? Also, with the iaxy's -- they should work (and ring) in Australia right? The only hint I'm seeing around is the use

In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here.... Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext => 700 ; What extension to dial to park parkpos => 701-720 ;