similar to: SIP powercycle not hanging up

A lot of people are going for the "VOIP only" approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com

I have a Sipura SPA-841. It's developed a nasty habit. At random times, it likes to dial my cell phone voicemail number and play my messages to anybody who happens to be within earshot. Any clues where to look at what's going on? My voice mail number (extension 220 in my dialplan) is the only number being dialed. When this happens, show channels looks like this: IAX2/NuFone-1

Anyone using IAX.cc / Sixtel? Would love to hear experiences good or bad. Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com

Hi folks ! I bought two sipura 841 phones. I used to have GN Netcom headset which I connect instead of the handset. The problem is that I don't have any sound coming out the headset and I can't speak neither ! I'am located in France and I was wondering if the cabling in the sipura and in the headset is the same (I mean the order of the cables) or maybe is there something else to

Hi there, I have bought 70 sipura 841 phones for a customer of mine. When following the mass provisioning guide in the admin manual for the sipura, I see it download the spa841.cfg file from my tftp server Sometimes the phone also downloads is phone specific file via tftp, and it works okay then. But, after a reboot of the phone, it is very very likely that it won't startup

Hello, I have spandsp setup to accept incoming faxes and receiving tif files via Email. Using tiff2pdf, or tiff2ps -a2 or even tiffsplit, the last page of the fax is cut off and the quality of the text looks "squished". I "figure" it's a tiff parsing thing, as opposed to a problem with my spandsp installation (heh). Has anyone experienced the same thing, or can

I'm wondering if tweaking [rx|tx]gain would improve my fax reception success rate. Running ztmonitor when receiving a fax shows 4 "octos" and an * on the RX side and nothing on the TX side. At the end of the page, there's a burst where RX goes to about 1/2 and TX goes to about 2/3 of the range displayed. Any opinions? Thanks in advance,

Is there any way to transfer a call from host to host and keep the call's variables intact? -- specifically, UNIQUE_ID and user created variables like CARD_NUMBER, EXPIRATION_DATE, and CVV2? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST Newline

I'm trying to pass a variable between servers using "setvar" in iax.conf. I have a box (ts2) with a t100p in it. It answers the call and dials another box (ast0) via IAX. I want to pass a variable along with the call from ts2 to ast0. I'm running CVS-HEAD-03/07/05 on ts2 and ast0. ts2's iax.conf: [general] disallow = all allow

I'm trying to spy on an agent (Agent/54321). I can "dial(Agent/54321)" successfully. If I "chanspy(Agent/54321)" or "chanspy(Agent)" all I get is a series of beeps. Any clue where I should start looking? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice:

I'm not receiving ANI2 (info digits) on my SBC PRI's. SBC said they're sending them. I called Digium support and was told it is not supported. Is anybody receiving ANI2 on a PRI? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST Newline pagesteve@sedwards.com

My boss is complaining that the delay between speaking and hearing in a meetme conference is noticeable and doesn't want to roll out our system until I can eliminate the delay. Personally, I don't think the delay is significant, but I don't sign his check. The system consist of 3 1u's, each with a single quad t1 card. Each card has 2 t1's running NFAS. The "t1

This is weird. If I have 2 members call into meetme using zap PRI channels on the box, they can here each other's keypresses. If I have 2 members call into a separate box using the same PRI's and then forward (dial(iax/...)) them to the previous box into the same meetme, they only hear a minor "squelch" for each other's keypresses. How can I completely mute a

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet loss, high jitter, etc manifested as "robot voice", dropouts, periodic

For those of you that are in the Portland, Oregon area: I am giving a talk today on Asterisk at the PLUG Advanced Topics Meeting. Details below. JT >From: "Zot O'Connor" <zot@whiteknighthackers.com> >To: PLUG LIST <plug@lists.pdxlinux.org>, > PLUG Announcement List <plug-announce@pdxLinux.org> >Organization: White Knight Hacklers >Subject:

Greetings! I was wondering if it is possible (using something like a group of Sipura SPA-841 IP phones) to have * announce information about the calling party via the SPA-841's speaker to a selected set of extensions that aren't set to "Do Not Disturb"... i.e., have * say the number, or perhaps have Festival speak the name, etc. If so, any hints and tips on how you'd go

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

I'm looking to purchase and deploy a bunch of hardphones for agent use. The phones will have to register with Asterisk and/or SER, depending on where the phones go. They need only one line, G729 codec, and no super fancy features. Preferrably something that is easy to provision. I would think the BudgeTone would be good, but then I've read so many people complaining about them, and some

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call