similar to: bayhamsystems.com experience

Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. Thanks, Doug. -------------- next part -------------- An HTML

We have tried Asterisk 1.0.9 on FC4 and have never been able to get CAPI (with Fritz card, fcpci) to work properly. Apart from that Asterisk works fine in switching internal calls. But it's useless if we can't make outgoing calls on our ISDN line. We are considering abandoning FC4 for Debian or SuSe. What is the general concensus on the best Linux to run Asterisk with CAPI? /Why Tea

Hi. Am new to this concept but have been requested to add VOIP capability to a small office phone system. They currently have 4 standard analog lines running into a PBX feeding 16 phones, with all the usual features, call transfer call hold internal calls etc. would the following seem reasonable ? asterisk server:- ( what specs ) cat5 > broadband (VOIP) 4 FXO's for incoming PSTN

Hi I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? Conrad

We've got a Toshiba DK system w/ analog ports that went to a voicemail server. I swapped in an Asterisk box with a Digium 4-port fxo card. It /almost/ worked perfectly. The problem is that Zap channels never hang up. They have to time out. I set up MeetMe, but all Zap channels hung forever. Very annoying. Same thing for FXO-to-FXO bridges. I figured out today why and fixed it.

Hi, I've had an inquiry for a small UK call centre, mostly outbound calls. I get the impression they are mainly calling 3G mobile phones, monthly phone bill, with calls is approx ?5,000 for several feature lines. How feasible is something like this with asterisk? I guess one big question is which type of circuit to use, ADSL in the UK is only 256kbs upstream, some providers do bonding but

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

Hi all, I've been working with * for a long time now, but only with analog FXS/FXO systems. I am venturing towards setting up a box in Germany now and I believe that requires a Fritz card? Do I even have to use the Fritz cards? Why not a Digium card.... We have 2 ISDN lines ( --> 6 handsets) so I'm guessing that will require 2 Fritz PCI cards (they have 1 port only). Then

Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt

Hi, I'm currently looking to run Asterisk in the office to replace an old PBX and would appreciate a little help. We are moving offices and will have 8 digital lines. My questions are: As there are 8 digital lines is this known as PRI? Which Digium card would be the best fit? Would you recommend looking at the echo cancellation cards? We are UK based: is caller id supported by Asterisk

Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Cheers, Richard. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060109/26c4a63c/attachment.htm

Hi, We recently tried installing Asterisk for a small office. We figured the safest way to go would be to buy from someone who sold equipment specifically for Asterisk and to use a consultant that they recommended. However ... it didn't turn out so great. Sound quality is terrible -- the echo is pretty bad, and there are popping noises, too. Callers say that people on the Asterisk end

Hi; I'm looking for a Polycom distributor in the UK who can supply a small number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ? jd -- John Daragon john@argv.co.ok argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas

Hi List, I am working on least cost routing code on the moment, and I am stumbling on a problem. Say you have provider A having: Prefix XXX 0.10 Prefix XXXYYY 0.20 And provider B having Prefix XXX 0.15 You're stuck, because you cannot decide if provider B's "XXX" prefix also covers XXXYYY numbers or not. If it doesn't, it would be a waste

I use IP Kall to forward my missed cell phone calls to. This way, if my phone is off, or out of a service area, calls will go to my * box. Concurrently, all incoming calls to my * box cause it to dial my local extensions at home, my extension at work, and my cell phone via NuFone. Problem: A loop can be created if my cell phone is not on. Say a call comes into my * box, it uses NuFone to call my

I know Vonage doesn't officially have a "bring your own device" type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060329/5bc9f644/attachment.htm

Hi Does anyone know a UK Voip Proivder that will give me more than 1 telephone number and point it to my sip account. www.SipGate.co.uk are great but they only allow 1 telephone number per user, you can register another telephone number by registering as another user but Asterisk doesn't allow multiple registrations. Many Thanks Scott

I am using Asterisk to send Voicemail out as Email. I am running into a problem I believe to be caused by the exchange server requiring SMTP authentication. I cannot get the sys admin's to turn it off. Does anyone know enough about sendmail to help me. I am assuming that the default mail client is sendmail. It will also send to other non-SMTP authenticated servers. Your help is much

Disclaimer: Not trolling. Cross-posting to -users to gague support. -users : Straw poll - if an XML based Manager Interface was avaliable as an option in asterisk.conf, would that be a good thing, or a stupid thing? >Have you ever tried initiating a session via XML with a terminal that >doesn't support backspace... I'm actually proposing that an XML I/F be avaliable as an option