similar to: * dials out zap line first 6 digits, pause, then last digit

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:

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

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

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

That was how I reset the black Iaxy I have used; I've never used a blue one. What I found was the initial provisioning would work fine, but if I tried to change the settings after having already provisioned the device, the provisioning program would hang, so I Googled for instructions on resetting the Iaxy to the factory settings. This was the procedure described at

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

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

> -----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, 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"

I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5" hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? -- Thanks in advance,

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

Hi, I'm playing around with llvm and the Kaleidoscope tutorial and first of all I have to say I'm really impressed. LLVM rocks! Unfortunately I've now run into a linker error while trying to optimize the IR or turn it to bitcode and likely due to my very limited experience with c++ I just can't figure out how to resolve it. The linker complains that the llvm::WriteBitcodeToFile

Hi, I recently purchased from a friend 2 Cisco 7905G for testing them with Asterisk. I was able to upgrade one of them with the SIP image, the other hang up during the upgrade process and now it won't boot again. When powered up, the red and green lights keep on and the screen is blank. Does any one know a procedure to fix this ? I do not have a contract with Cisco, I have even call a

Hey All, Having a bit of a problem with a Wildcard X100P card. When I try to make an outbound call using the card, it picks the line up and then only dials a single digit. I've confirmed it's only dialling a single digit by listening on a phone plugged into a parallel socket. Incoming calls work fine, it's only outbound calls I'm having a problem with. I've tried

I have a CentOS-5.6 remote server in a house in Italy, where there are occasional thunder-storms. There was one yesterday, when the electricity went off 3 times, for a second or so on each occasion. My server, an HP MicroServer, came back (re-booted) on 2 of the 3 occasions, but not on the third. I assume that the problem arises because the machine does not close down properly. (Although it is

Hello AstLinux seems quite suited for my use. Can you configure more incoming port via a web interface? I'd like to install it to a "normal" hdd. Can that cause any problems? BR Amund Nygaard -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Kristian Kielhofner Sendt: 4. juli 2005 03:23 Til:

I committed a new version of service.c and daemon_test.rb. Now you can pause and resume a Ruby service module. Regards, Park Heesob --MIME Multi-part separator--

Hi all, While testing the pause/resume functionality with the daemon_test.rb script, I noticed this odd bit of behavior. It appears that sometimes, the pause command seems to actually stop the service. It doesn''t happen every time - just sometimes. Here is a bit of command line history to demonstrate: C:\eclipse\workspace\win32-service-0.4.0\test>ruby daemon_test.rb start VERSION:

I used pause() from library(DAAG) to pasue between plots. This works when I source a script, but seems don't work when I run (ctrl + R) the script in R. Did I do something wrong? Thanks.

I used pause() from library(DAAG) to pasue between plots. This works when I source a script, but seems don't work when I run (ctrl + R) the script in R. Did I do something wrong? Thanks.