similar to: Routing calls by trunk?

I have read through the info available on the list and searched for things with google. I have found quite a bit of info on the subject, mainly to do with 'simply add this to iax.conf and this to extentions.conf and it will work'. So I tried all of those things and I can't for the life of me get this thing to work properly. What I want to be able to do is dial ext 2009 from server

Hi, I am just trying to figure out how to setup a TDM04B card for incoming/outgoing calls. I have 4 lines, which are provided as a rotary trunk group, currently hooked into a Nortel system, which asterisk will replace. I have setup a Dell 1800 (Tower) system with the TDM04B card, which seems to work. The question is how do I set it up that all 4 lines are part of a trunk group, such that

Hello All, I have new TDM04B installed and working fine with Asterisk 1.0.5 built on RedHat 9. All is working fine except CallerID that bothers me big time. I have several Panasonic and Sony phones and CallerID works fine with it (when I plug in the line into phone instead into Asterisk I get CallerID) but fails with Asterisk. I am based in California (San Francisco) and my Telco is SBC

I'm installing my first Asterisk server. I have a TDM04B card installed in my asterisk server (4x FXO ports). I have 5 Cisco IP phone 7960 working fine on asterisk using SIP. My configuration to receive call is working as expected meaning anyone calling on one of the 4 FXO ports is answer by asterisk and asked to enter the extension of the person to reach and then it is transfer on the

In the next week or so, we'll be turning up a new T1 Pri from Cox Cable in Omaha using a T100P (installed but not yet configured). While discussing the interface parameters with a very knowledgable Cox engineer, we decided on some of the basics including b8zs, esf, dchan=24, callerid, 5 digits of called number forwarded to *, and switchtype=national (NI2). They indicated their CO switch is a

I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is having calls dropped. Sometimes you can stay on the phone for a long time and sometimes the call is dropped within a minute. There are 9 lines connected to 3 TDM04B cards. The Panasonic Pbx we replaced did not have this problem. There are 8 SIP phones and 16 analog phones connected to two Astribank-8 units and everyone

>All, >I have a TDm04b card an 8 SIP extensions. Calls come into the TDM and >are answerd by the auto attendant. When an extensions is entered I see >the Dial(SIP/100) on the console but the phone never rings... >I can pick up any extension and call 100 and it rings just fin>e and they >answer and everthing is fine. All 8 extensions can dial 9 for an analog line >and call

I have installed Digium TDM04B with the latest CVS. However I have encountered following problems: 1. When it dials out, many times the digits are not properly recognized by telco as I hear the announcement "please check the number and dial again" although I see on the screen that the dialed number is correct. 2. When the call is forwarded outside, with something like exten =>

I have an Asterisk server that was running fine until Sunday. Monday there was a power outage and the server was off most of the day. This server has a TE110P, two TDM04B and an Astribank 32. Today I noticed that the TE110P started having IRQ missess. Before today it only had about two or three per month. Today, this is the miss count after ten minutes on: IRQ Misses: 1443

We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a "you must first dial a 1 or 0" message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

I just install a Digium TDM04B card. I created 4 separate Zap channels and one outbound routing containing zap channels from 1 to 4. If a phone line is plug in Zap/1 then works fine, but if I unplug phone line from Zap/1 (simulating fail) the system keep dialing out on Zap/1, even with no dial tone; Only if Zap/1 is busy it will use Zap/2. There is any work around or different setting to avoid

hi! I have a question of ISDN PRI "calling number" setting for outgoing call. I need the receiver to see the CLI that I have set. Eventough I've set the CLI as below, the receiver keep on getting a fixed number( used with the PRI to receive incoming calls). My ISDN PRI E1 provider says he has not done any restricions on the custom CLI side.

Hello, I have installed asterisk version 1.2.12 and latest zaptel modules. but i can see some IRQ conflicts on the server. iam uisng two TDM04B cards. according to my previous knowledge on asterisk verison 1.07 asterisk has given lot of erros when starting if you have assigned the same IRQ number to any other device. My question is new releae version 1.2.12 has resolved the IRQ issue ?? TDM04B

I am having problems with * not hanging up an incoming PSTN line, if that line is not answered before the person calling in hangs up. The line hangs in various states, it has hung with a busy tone, with no tone at all. I am running *@home and have a digium 4port line card. This was configured by the genzaptel command I then added trunks for each line. I also have a Pulver WiSip phone which I

I have a customer that is using a digium tdm2400p card with multiple analog lines coming in. They want to be able to forward all calls on specific trunks to specific ring groups. I tried this with the inbound routing and specifying the phone number of the trunk and directing it to a ring group but this doesn't work. Maybe I'm going about this all wrong. Any assistance would be appreciated.

I'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is

I have TDM40B, TDM04B cards, 4 analog and digital phones. First I want to use 4 analog phones with my TDM40B card. I would like to dial between 4 analog phones. The dialing numbers for 4 analog phone will be 800,801,802 and 803. These are my conf files. /etc/zaptel.conf fxsks=1-4 fxoks=5-8 loadzone = us defaultzone=us ;;;;;;;;;;; /etc/asterisk/zapata.conf [channels] relaxdtmf=yes

Hi, I've been lurking here for about a month and I've been putting together our companies planned migration to a new office and a new phone system. Could anybody tell me if there are any pitfalls to the following setup I may have missed? 1: TBD PC running * a: Digium TDM04b FXO connected to PSTN 1: lines rolling over via SBC sentrex to get multi-lines on 1 number b: Digium

Hello, I'm planning to add Asterisk as VoIP gateway between two existing offices, A and B. A has a DCS Compact II PBX with a spare 4BRI card which I plan to connect to an ISDN card, probably an EICON Diva Server 4BRI) on my VoIP gateway. The gateway will be connected via IP to the B office's gateway. B has a DCS Compact PBX without spare lines. To handle this, I thought I