similar to: Invalid Extensions -- More like traditional PBX systems?

That could explain why it wouldn't work on any of my sip extensions I tried it on this morning when I first read about it and thought cool the things you learn. Is there anyway to make it work on Sip extensions? Cheers, Dean -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Aaron J. Angel Sent: Wednesday, 23

Hi everybody, I'm running an asterisk box -cvs version since few monthes, updated it middle of may and a last one on thursday (24 june) Since this one, my H323 calls loose they audio, both sides. Calling directly from Gatekeeper is ok, so problem comes from h323 asterisk channel. I saw few people telling about similar problem begining of month, does they solve their problem? I also grab

Hello I have heard that i can put a file in a certain directory to get * to initiate a call. Is this true ? if so where would i look ? Best Regards Simon Garvey

I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki mailboxnumber@context ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a

This is somewhat related to my other query on the list regarding NAT traversal. I have heard many times that IAX is "NAT-transperant". I am unsure how it accomplishes this. I do know that SIP works like this: your SIP device send a request to the SIP server (usually on port 5060) with whatever command. The SIP server respends to your device's "apparent" IP and port (this

Hello. I am looking for a simple Asterisk billing solution. I expect about 50-100 users (a mix of IAX and SIP) through 3-5 outgoing providers (all IAX). I need something that can handle monthly fees and per call charges (depending on destination, obviously), and should provide a web interface for customers and administrators. Something that can tie in to one of the existing management GUIs

Hi, i have a strange situation: A domain is still running where domblklist points to a snapshot file and also dumpxml says the current drive is that snapshot file. But the file has been deleted hours ago. And the domain is still running. I can login via ssh, the database and the webserver are still running, domain is performant. How can that be ? Also lsof shows that the file is deleted:

Peter, Yes that is a possibility. I will try disabling PAM (or switching the auth order) and see if that makes a difference. Thanks for the suggestion! ~ Laz Peterson Paravis, LLC Ph: 951.319.3240 x201 > On Jul 1, 2015, at 11:34 PM, Peter Chiochetti <pch at myzel.net> wrote: > > Am 2015-07-02 um 01:41 schrieb Laz C. Peterson: >> >> I did attempt to switch the

Dear all, I'm currently running R on my laptop -- a Lenovo Thinkpad X201 (Intel Core i7 CPU, M620, 2.67 Ghz, 8 GB RAM). The problem is that some of my calculations run for several days sometimes even weeks (mainly simulations over a large parameter space). Depending on the external conditions, my laptop sometimes shuts down due to overheating. I'm now thinking about buying a more

Thank you for the response Axel. I will look into that. I did attempt to switch the PAM/Kerberos authentication to Dovecot LDAP authentication, but now performance is unbelievably slow. For example, with PAM/Kerberos, a user can log into webmail and have all of their emails/folders showing almost immediately. When using Dovecot LDAP, it takes literally 8-10 seconds to see the same thing. I

Hello, We have a vacation/mail forwarder plugin in squirrel mail that allows customers to forward email. It has come to my attention that there is a limit (appears to be 5) on the number of addresses that can be specified. I'm now trying to track down where this policy limit is set. This is the error we see in the customers .dovecot.sieve.log file: main_script: line 26: error: number of

Dear R gurus, I want to wrap labels that are too long for a plot. I have looked at strsplit(), substr(), nchar(), and strwrap(). I think it's some combination but I'm having difficulty trying to figure out the right combo. I think I need to create some new matrix containing the labels already split, though I'm not sure if maybe there is a quick and dirty way to address this

As kindly pointed out to me (oh my decaying gray matter), is.object() is better suited for this test; $ svn diff src/library/base/R/diag.R Index: src/library/base/R/diag.R =================================================================== --- src/library/base/R/diag.R (revision 68345) +++ src/library/base/R/diag.R (working copy) @@ -23,9 +23,11 @@ stop("'nrow' or

I'm not sure if this is possible, but I was hoping to find an address book that runs on Windows XP that will allow me to select a phone number and send that to my Asterisk. The Asterisk system would make the call and connect the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out there that can do that? Thanks, Dave -------------- next part -------------- An HTML

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either

Thanks for that. I just got rid of packet 8 and went with 100% asterisk in my house. But I use the PAP2-NA and RT31P2 from Linksys for my FXS ports. But would like to have an extra FXS laying around just in case.. .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From:

Just upgraded to the current stable ver. when I start asterisk with -vvvvvcg I get the following error [pbx_loopback.so]Dec 23 19:25:33 WARNING[1633]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_loopback.so: undefined symbol: pbx_substitute_variables_varshead Dec 23 19:25:33 WARNING[1633]: loader.c:440 load_modules: Loading module pbx_loopback.so failed! Asterisk

Could someone please direct me (via personal email) to a provider with good prices on Polycom Soundpoint IP 500's with POE cables? I need 14 of them. Thanks, Adam ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody Road, Suite 380 Atlanta, GA 30328 Office: 770-395-0088 x34 Fax: 770-395-0989 Mobile:

I've been searching the forums and on the list to see if this has been addressed. If it has, could someone point me to the thread to fix or at least acknowledge it is an issue and what is causing it. Posting to the list was last resort as I couldn't find a solution anywhere else. Setup: Asterisk@Home 1.3 (this is my first system, so path of least resistance) Digium TDM400P (2 FXS on ports

Hello. I have an * server set up on a public IP. I have SIP clients at three different locations, all behind NATs. I have all the SIP users set up this way: [user1] type=friend username=user1 secret=password1 callerid="User 1"<101> host=dynamic qualify=yes context=outgoing All three SIP clients are configured to use STUN (using stun.fwdnet.net:3478). Furthermore, I have