similar to: IAX connections broken

Just a quick listing of tested, and updated, steps from my notes. Enjoy ! http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html -baji. --

Will this work to match any number from the 770,404, or 678 area codes? _[404|770|678]NXXXXXX If this won't work, is there a pattern that will do this? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> Attachment encrypted? click here <http://www.highpoweredhelp.com/tutorials/wincrypt/> .

I am trying to use PHP to reload the extensions in an Asterisk installation. I keep getting this error: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) when I run the script by visiting the URL; however, if I run the script from the command line, it runs just fine (works perfect, actually). I think it is permissions related. Does anyone have any ideas? <php

Is there anyone interested in developing an open source Asterisk / MS Exchange solution? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> Attachment encrypted? click here <http://www.highpoweredhelp.com/tutorials/wincrypt/> . -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 ;

Polycom Provisioning Tool Updated. I made a bug fix that was reported, which was causing the directory creator to not work when there was an invalid character in the filename of the csv. I have also posted an FAQ: http://www.wintrisk.com/ppt.html#FAQ Download the new one, and tell me what you think! It's free, and mildly useful! http://www.wintrisk.com/ppt.html Yours, Michael Munger,

hello, running asterisk 1.4.11 on CentOS 4.5 I am getting no response on function STRPTIME() the system just hangs, STRFTIME() is working fine as seen below. Same thing happens whether I called in from a softphone or via teliax. While executing the following code : ; exten => s,n,Set(v_ts=) exten => s,n,Set(v_ts=${STRFTIME(|America/New_York|%Y-%m-%d)}) exten =>

We have an asterisk system and Polycom phones that were provisioned by someone else. They want to get call waiting to work, but for the life of me, I cannot figure out why the Polycom is returning a SIP 486 Busy Here when you call and the person is already on the phone. I have the feeling there is a configuration in sip.cfg or mac.cfg that I am overlooking. Any thoughts? Calls per line key

May be slightly off topic, but I was wondering what everyone thinks of this latest ruling against Vonage? Does anyone really know what Verizon hold patents for, and could those patents possible affect anything in Asterisk? Who knows who Verizon will go after next. Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

What modules do you want on it? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of MOSBAH ABDELKADER Sent: Saturday, August 04, 2007 3:16 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Connecting

Hi List; What is the difference between WaitExten function and TIMEOUT (response)? As I see that both are used to determine the allowed time to enter the digits, any one can advise? Regards Bilal ____________________________________________________________________________________ Shape Yahoo! in your own image. Join our Network Research Panel today!

I have a phone (a panasonic globalrange phone) which always sends a fully qualified phone number. That is, for a local Canadian number, even if I key in 6135551212 it actually sends to asterisk 01116135551212. This means of course, along with "normal" phones I end up having twice as many extensions for outdialed numbers. Is there any way I could canonicalize this down to the more

I just changed the from DSL to Shaw Cable (static IP) configure the firewall but now asterisk I can not register with FWD nor VoipJet calls going out. I am using IAX with FWD Did I missed to change a setting? I don't think there is any though. I am on shaw extreme connection; I just talked shaw tech. and they are not blocking any port - I was told. So why IAX will not register with FWD and

I'm trying to find the right Digium card for the Dell 2950 Dell 2850 HP DL380 G3 HP DL360 G3 Are these 3.3v or 5.0v machines ? I am out of the office, and need to buy a card today. I am looking at either the TE407 or TE412, and would appreciate any help. :) Julian

Stealth Communications Announces Registry to Avoid Access Fees Posted on: 04/23/2004 Stealth Communications Inc. today announced the official launch of a registry that allows service providers routing calls over the Internet to avoid paying local phone companies access charges. The VPF ENUM Registry allows carriers to map telephone numbers to IP addresses for such things as SIP phones and

I think you need to increase verbose output and search in /var/log/asterisk/full for any error message related to IAX2 registration or simil. 2017-06-05 17:12 GMT-03:00 <thelma at sys-concept.com>: > No, I don't think it is IP table issue, I've not upgraded dd-wrt for a > while and it was zoiper was working OK with my previous version of > asterisk. > > After upgrade

I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it

Hello Instead of using PrivacyManager, I'd rather use my own dialplan to prompt the user for a ten-digit number if they called while blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid"

Just got one of these. Horrible to program. Trying to key in the FTP server. Won't even remember the info after rebooting. Anybody know the proper way to beat on this stupid beast so it will work?

I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569