similar to: Question on setting up asterisk with hunting lines

Hi, Need to implement hunting (create a hunt group so my subscribers can have a single GSM number for access to me)of GSM SIMs on a GSM bank independent of the Telco for the SIMs. Anyone got an EXACT idea how to do this? Thanks, Latex. ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football

I was wondering two things: I am running Asterisk Current Stable. Is there a way in Asterisk or using other equipment or through the TelCO that I can make a channel bank such that say I have 3 Phone number either with the use of FXO Cards or by using 3 connections via IAX to a service that could then provide PSTN connectivity where when Line 1 is called the call is then switched and held on

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

You should be able to get ISDN though. I think BT provide this anywhere. Otherwise your solution will be a Primary rate (ISDN30) with 8 channels. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matt Sent: 07 June 2004 10:19 To: asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] Multiple DDI &

Hi everyone, I want to get multiple DDI's and hunting across those DDI's in case one of the lines is busy using analog phone lines. The system is for a large house so I want 3 x PSTN lines. 3 x DDI's and the ability for those DDI's to be presented across all three PSTN lines. BT say you can't have more than one DDI number associated with a PSTN line and that you can't

Hi I am trying to develop a night divert. Caller dials in after hours on Zap and it gets divert to a mobile number via a second Zap. The call bridges but will not hangup the channels when the parties finish. Is there something I am missing or an dial option that I should be using. I am using latest CVS. [night] exten => s,1,Answer exten => s,2,Wait,1 exten =>

Hello All, Has anyone configured bri to answer for only one msn? In essence, when the primary is busy I want to have channel 2 ring. I am using an eicon diva server bri I know I saw it in the windows interface, but don't see it in the linux setup. Regards, Greg

Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Thanks,

I've tried to transfer a call using the Flash command, but with my configuration it doesn't work. I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR: TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk >From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can

Hi All, We recently had an old office building burn down. The office housed maybe 20-30 people. Only about 10 or so of those had their own extensions. We had a standard pbx from an area communications company, and I'm not quite sure about what kind of phone lines were there, I only know that their were actually 3 phone numbers, but everyone could get an outside line if they needed

Hello, We have a high volume of incoming and outgoing calls that come in via our analog POTS lines connected to FXO cards in an Adtran TA750. This is connected to a T100P. We are using Polycom IP 500's. The problem we are experiencing is, on frequent occasions, when someone dials out, there is another person who has just dialed in. I have had this problem at my house before (without an *

With multiple inbound PRIs (and hunting across them) coming to multiple [asterisk] servers it is important to be able to do administration, i.e. control which PRIs in the same hunt group take (and which don't take) calls from telco at any given period of time. Our pre-asterisk platform uses SERVICE commands for this purpose to put B-channels into 'out-of-service'/'maintenance'

Hi all, We have an analog line from telco, on which 3-way calling is subscribed to. This line is plugged into an FXO module on a tdm400p. If an incoming call comes in on this line, can */zaptel send Flash to telco via the FXO module? If it could, then an incoming call could be 'transfered' to a cell-phone, for example, with a single analog line. (where 'transfer' is really

I'm trying to figure out if I'm the only one with these problems with them. I recently had a few customers that switched to them because of the price, of course that means that they have to use FXO ports, but it is realy cheaper, so customers don't really care. In any case, there are 2 issues that I can't get solved, and they are not interested in helping. 1. When they tell you

[Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones "inside?" [In other words, I obviously need a NIC -- but what would be the

I have an interesting dilemma with sequencing in the dialplan. Up to now, I have assumed that the extensions in the dial plan were tested in the order that they appear in extensions.conf. In other words, I have the following fragment which was designed to dial toll free on the PSTN and all other long distance on VoIP: [longdistance] include => local

Hi I have a TDM card with 4 lines on a hunt group coming in. I can answer the phones with exten => s,1,Answer() exten => s,n,Dial(Zap....&Zap....) ... The problem is I don't know how to find out what extension was originally dialed. And, trying to match on the extension always fails. E.g. exten => 1234567,1,Answer() # never gets here I thought I could get the extension on

Hi all, I have an existing panasonic analog pbx in use and a asterisk server with digium tdm400p(2 fxs and 2 fxo). channel 1 -> fxs -> telephone channel 2 -> fxs -> telephone channel 3 -> fxo -> extension 15 at panasonic pbx channel 4 -> fxo -> phone line from telco We call in to fxo (channel 4) and enter the ivr which prompt us to enter the extension number. After

can anyone recommend a plain ol fashioned telco in north america where i can get a DID with call hunt and voicemail which can be programmed via web interface. thanks and pls pvt joe

I'm having a major problem at one of my branch offices with "Phantom Rings" on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and