similar to: Polycom SIP Phones

I just set up a Polycom 500 on *. Every few calls I make, the call connects and the receiving party can hear me (thru Broadvoice), but I still get ringing on my end, as if they never picked up. * logs look just fine. Does any one have any suggestions? Thanks. ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody

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

Could any of you provide me information on a good Polycom phone dealers to utilize. One who provides firmwares ..etc Thank you! Kenny ______________________________________________________ Click here to donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/

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

I am attempting to update my Asterisk installation from 1.0 to the latest stable version. When I use CVS checkout, I am receiving the following messages on chan_sip.c: RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.510.2.25 retrieving revision 1.510.2.27 Merging differences between 1.510.2.25 and 1.510.2.27 into chan_sip.c M asterisk/channels/chan_sip.c Then, when

I was wondering if there was a way of setting up the dialplan in a way that if you dial an extension that is NOT in the dialplan then it would play a not-in-service gsm file and then play congestion tones. I would rather like this better than just hearing a busy signal on my phones.. I DID search around on the wiki and using google and could not find anything. Thanks. -- Stephen Rosebush,

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

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

Has anyone ever triend to hang a eSATA drive chasis off oc a centos box? Not specifically this box, but something like this.... http://www.compsource.com/ttechnote.asp?part_no=DS1220&src=F I have used internal SATA drives on 3ware cards before without an issue. Didn't know if this was in the same arena. Thanks! Andrew

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

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

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

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

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

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:

I am looking to help out my company find a more budget conscious but reliable way to hold conference calls between 5+ people. 4x a month we hold several hour long conference calls during non-business hours. All of the employees have high speed internet. Currently we dial up an AT&T conf using regular analog phones. I don't have a great grasp as to what Asterick is capable of, but my

G'Day All, How do I fix this: I receive a call at the extension. Press the hold button. Music on hold starts. When I place the handset back on the cradle, the call gets hung up/disconnected. The Phone is A GrandStream Budge Tone 100. Thanks

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

Hi, I've ordered a few IP501s from PC Connection, basically since we have an account with them. I like the phones for what they do, and now would like establish a relationship with a reseller that can give us maintenance and access to the most current firmware. What are some good resellers out there? Regards, --- Gavin Adams VP of Technology Promisant (USA) Inc. Email: me@gavinadams.org