similar to: Cell Phone, *, Portability

I'm presently working on an office move and evaluation of telecommunications services needed at the new location. I'm presently wrastling with an issue related to portability and geography between landline carriers. Presently certain people within the organization are hopelessly in love with our 909-822-xxxx number(provided by pacbell/att). As that number is presently provisioned it

Hi Folks, I recently setup an asterisk system in order to provide a telephone phone system for my web hosting business at a very low expense. My problem is that DTMF tones are not being recognized when calling the IPKall phone number. Calling my server via FWD and IAXTel works out fine however. Has anybody experienced this with the IPKall service? are they not passing the DTMF tones or am I doing

Before I got into my question for the day I'd like to applaud all the helpful folks and time spent behind the asterisk project to get it where it is today. Great work and between this list, the doc list and the irc channel it's been a pleasure to deal with people willing to help others when and if they can. I will be moving in a few months and I'm concerned as to what kind of

https://bugzilla.mindrot.org/show_bug.cgi?id=2596 Bug ID: 2596 Summary: Brother printer (((1~855~709~2847))) Technical Support Phone Number Brother Printer Customer Care Phone Number Product: Portable OpenSSH Version: 4.0p1 Hardware: Other OS: Other Status: NEW

https://bugzilla.mindrot.org/show_bug.cgi?id=2597 Bug ID: 2597 Summary: Epson printer (((1~855~709~2847))) Technical Support Phone Number Epson Printer Customer Care Phone Number Product: Portable OpenSSH Version: 4.0p1 Hardware: Other OS: Other Status: NEW

want to get from SIP to ISDN or from SIP to SIP. I only have a ADSL connection that means 786kb/s downstream and 128kb/s upstream so i can max handle 2 sip calls at once. I want to have 25 Accounts because of the different numbers for the different phones. Good, i wanted to buy a ready built pc from ebay but now i think i will built a 19" rack case so i will built in a 1 - 1,5ghz intel

Hello! I am semi-new to asterisk, I've been toying with it for about a month now. I'm using the current version from the CVS server (as of last week) and have succesfully connected it to Broadvoice, as well as a few softphones. I got a Nortel i2004 IP phone from a friend to setup at home as a homephone using Broadvoice, but going through my * box and using iptel.com for toll free calls.

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

https://bugzilla.netfilter.org/show_bug.cgi?id=1080 Bug ID: 1080 Summary: call ^^18557^O92^847^^@@ Epson printer customer care phone number Product: bugzilla Version: other Hardware: x86_64 OS: All Status: NEW Severity: enhancement Priority: P5 Component: netfilter

https://bugzilla.mindrot.org/show_bug.cgi?id=2616 Bug ID: 2616 Summary: quickbooks phone support ,quickbooks support phone number,quickbooks help 1877 778 8714 Product: Portable OpenSSH Version: 7.3p1 Hardware: 68k OS: Mac OS X Status: NEW Severity: enhancement Priority: P5

Well I know from personal experience that NuFone is working on a solution for its customers as fast as it can. I know they found an alternate termination provider and are working to have a solution for the TF and Local DID's he currently has on his platform. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

Ok so I'm thinking about migrating to a Flex T-1 with 1.024Mb Data and 6 Hard Incoming/Outgoing Phone Lines. I can either break them out via channel bank and use a FXO to convert them but I'm told it will add a dely of at least one ring. And you have another analog/digital conversion to the mix. The 2nd option is to bring them in over PRI's Question is the solution provider will

I understand that softphone are the answer in fact I deploy a ton of the Ip comm version every week. I am under contract with the phones so I can't sell them and there no easy way out of the contract. As for 79XX's I have several office that have them working over a VPN backed in to our main office where the CCM's and GW's are with managable problem and for the most part they

Well this is disappointing. Time to find somebody else... -- Wes -----Original Message----- From: NuFone Operations [mailto:support@nufone.net] Sent: Tuesday, April 18, 2006 3:44 PM To: wbaehr@totalmac.net Subject: NuFone Update: DIDs Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier supporting the Toll-Free and Michigan DID operations of NuFone, has threatened to terminate

Having had various issues with local vendor (begins with "V"). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? >From what I've read, the Fed's seem to say "no", they cannot refuse, or impede this. joe a.

Hi, I have had an issue for a long time, and really just can't solve it. My boss and others seem to have a problem when they call into our asterisk phone system. It often takes 3-4 tries of entering an extension before the system gets it right. Below is my context that the call comes into, and some debugging from the asterisk console. [corporate] ;;exten => fax,1,Macro(faxreceive)

I'm still a newbie in Asterisk, just yesterday installed it for home use (so I can call home while travelling). Using AVM A1 (BT Speedway) ISDN card. Anyway, I find it very hard to locate supporting information for Asterisk. User's Handbook is still a draft, this mailing list is not easily searchable, and any info out on Usenet is scarce. I spent some 30mins just trying to find out naming

Calling 7777 from a local extension on my local network, I get good voice quality from asterisk, and asterisk reliably recognizes my dtmf input. I set up a sipphone trunk (free) and called in to it via a separate sipphone account on another computer, and got slightly lower, but still good, audio quality. I set up a FWD trunk (free) and called in from the other computer, and got somewhat lower

I didn't get any response at all to my last "request for status" on IAXTEL. So, when this happens, I attribute it to one of a number of things: 1. No-one knows. 2. No-one cares. 3. Everyone knows, but are too busy to reply. At any rate, my investigative side kicks in and I began searching thru the digest's I've gotten, looking for references to IAXTEL. Mostly it is

<BLOCKQUOTE style="PADDING-LEFT: 8px; MARGIN-LEFT: 8px; BORDER-LEFT: blue 2px solid"> <DIV>Hello,</DIV> <DIV>&nbsp;can someone tell me what's wrong with this? I can't make toll free calls via iaxtel. Here's the definition in my extensions.conf</DIV> <DIV>&nbsp;</DIV><FONT size=2> <DIV>[iaxtel-trunks]</DIV>