similar to: callers who don't press any keys

Folks, I've got one of those GPO 1950's rotary dial phones that I'm trying to get working in the UK. I've got pretty much everything working with my TDM400, the phone rings and I can receive calls but I cannot dial with the rotary dialer. I have set pulsedial=true or whatever the exact setting is and I can dial from the phone by lifting the receiver and tapping out the number on

Will the FXS cards that work with asterisk handle rotary? Are there any channel banks that can convert rotary to touch tone (like some sorta bridge)? The goal is to be able to log input from rotary phones. Full PBX functionality would be nice but... (It's for a project, not for serious production). -- // Ethan O'Toole //

I'm trying to use a couple of old Western Electric type 500 phones (desk model, rotary dial). These phones work fine, as tested with telco lines (they dial, receiver/transmitter works fine, etc). I'm running Asterisk 1.6.2.11. I can't get them to dial through Asterisk. They are connected to a Rhino channel bank which is connected to Asterisk via a Sangnoma card (T1 with echo

Hi. Does anyone know about a simple one-fxs ATA with pulse dialing support that can work with Asterisk? A SIP one would be ok. I've been told that the Digium S101i IAXy does support pulse dialing; although it's a iax2-only ata it could be enough. I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards

Ok... I am not a telephone guy... I was born after rotary phones, so forgive my ignorance in this matter. I am trying to get a really old rotary phone up and running with an ATA. Why? Who knows... just thought it would be cool. The problem is that it does not have an RJ11 connector, instead it has three wires (green,yellow,red). Does anyone know what that type of connector is called? Or

On Tue, 2005-01-18 at 09:49 -0600, asterisk-dev-request@lists.digium.com wrote: > > Hi, > > I am Arnaud F?vrier, I teach in a technical university in Marseille. > > I'd like to know if is is possible to connect a very old phone to > asterisk and dial pulses with it? > > Are digium cards pulse dial compatible? > > Is there any specific configuration

I am trying to download the latest release of Asterisk from: ftp://ftp.digium.com/pub/asterisk/ The link provided by Digium is incorrect for the Asterisk Tarball as there is no such file at ftp://ftp.digium.com/pub/asterisk/asterisk-1.0.7.tar.gz However the links for the Asterisk-Addons and other Tarballs is OK ftp://ftp.digium.com/pub/asterisk/asterisk/asterisk-addons-1.0.7.tar.gz Does anyone

Hi, Long story short - I have an ancient Britsh Telecom phone attached to my Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call quality is excellent. However, dialling out is impossible, as Asterisk consistently mis-reads the number of pulses the dial sends (it could be a squiffy dial, I'm not sure). Not to mention the fact that, in today's modern "want

These taken tonight (9/22/2004) at the Expo and Reception Enjoy. http://photos.tropiano.org/gallery/astricon-2004 Lenny

Hi I was wondering if there are any special settings that I need to be able to transfer calls. Whenever I press the 'recall' button, I just here a click, and no ring-tone to transfer. in my debug log I get this : -------------------------- Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1 (index 0) Feb 24

Hello. I'm trying to transfer calls from an analog phone (Zap/1, TDM400P card) to some other terminal connected to my Asterisk PBX. If I make a flash hook pressing the phone hangup button quickly it works as expected, I get a new dialtone and the other side is put on hold. But I would like to use my phone's "R" key instead for some different reasons (it's quite easier to use

When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this.

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Hi all, I have the following setup PSTN -> ASTERISK -> IVR (using dialogic card) 1) Caller id information is presented to asterisk during the first and second ring. 2) Hence, Asterisk waits for 2 rings before pickup the call and forwarding to the appropriate FXS port. 3) The IVR application also waits for 2 rings before picking up the call to get the caller id. 4) Hence any caller

Hi there, I'm experiencing an echo problem and dammed If I can sort it out. We're running Asterisk on Fedora Core 3 64bit, installed as per http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3. These are the specs of the Machine ? 1 x AMD A64/3500+ CPU: Desktop Athlon64? Retail w/fan SKT 1 x Asus A8N-SLI Deluxe Athlon? 64 S939 NVIDIA nForce(r)4 SLI? PCI Express Req: 24pin ATX 1 x

Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off

Hello all, We have a slight issue to resolve. We have a client who we are drafting an SLA for the delivery of telephony services using Asterisk. Nothing extraordinary. However, we do need a way to measure our service availability. We currently use Nagios and Cacti to monitor server availability as well as asterisk and mysql responsiveness, and last, "ping" availability to our

Hi, Has anyone had success using a TE410P card in an HP-Compaq DL380 G4 server? For me the card is detected fine, but the system just never sees an interrupt from the card. I've tried everything I can think of. The card definitely works. Its Fedora Core, but we also tried a stock 2.6.10 kernel. We tried with and without Hyperthreading, with "noapic", we disabled all the

Folks, I have a TDM400 with an FXS module. I'm having some trouble ringing some phones attached to the device. I have a real el cheapo handset that cost me about 9UKP. It works fine, I plug it into the Wildcard and call the channel and it rings. I have tried plugging in my BT Diverse 3016 cordless phone but it will not ring, although I can call out from it. I have also tried my GPO 332L

Hello there! I'm trying to get sound from applications running from other users bug the one who owns the current GNOME sessions. Typically, my default user is "A" and he's running the GNOME session, logged in graphically. From this session, I open terminals, su to other users (B or C, non-root) and run mplayer or firefox. No sound for these. Adding those users to the