similar to: yet another question on DID trunks

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something that'll work? Thanks, -Ken

What they are probably marketing is putting in their own equipment out there. I install a product that does exactly that. A paradyne jet fusion. It takes care of the part of which channels are data and which are voice. If it's anything like these, the lines will come out on pairs. You will then have to use channelbank and FXO/FXS cards to get it into your phone system. The jet Fusion

Hello, I new to this, but with the help of mailing lists archives and IRC I am able to build my PBX. Thanks to all who had help me to reach till here. I am stuck at a point where I can't find the solution on mailing lists or even on IRC. I have individual 4 DID (Direct Inward Line) coming from Telco and terminating into TA 750 to FXS card. Many of them told that Phone instrument terminates

Hello everyone, I'm hoping someone can help me with this. I have a business customer in the U.S. (Michigan, AT&T Territory). I need to get 4 trunks into an asterisk Box. My intention is to use an Eicon Diva Server card with 2 BRI Circuits. The reason for this is that the business needs DID's on the trunks (20 of them). A full or fractional PRI is overboard for them, as they will

I am currently using a T1 PRI from TWTelecom for DID and outgoing calls, but I recently discovered that they're offering call termination/origination over SIP trunks in my area now. If they could deliver these SIP trunks to me over a guaranteed-QoS circuit, this would be of great interest to me. We're already using a DS3 circuit from TW for our internet uplink, so I'd imagine it

I wrote a week or two ago about a problem I was having with a TDM400P with 2 FXS. The problem was reliable fax transfers. There were often slow or had communication failures during transmission. Richard replied to my post and told me this is an ongoing problem that might have a solution in the weeks to come. However, I was wondering if anyone had any alternate solutions for faxing with an

We have 4 analog line and 2 analog trunks. On the trunks we have all the DIDs coming into the current phone system. Trying to get everything moved over to Asterisk but having issues picking up the calls on the analog trunk. We can receive calls on the plain analog lines and we can call out on all analog lines and analog trunks. When a call comes in on the trunk line the ZAP channels don't

Hello, I am having some problems with faxing in asterisk. I have a TE100P which is taking my PRI. This seems to be working fine. I also have a TDM400P with 2 FXS. Again card seems to be working fine, I can dial from phones attached to these to ports and everything seems to work fine. I have 2 DID's for my two fax machines that dial the Zap port for the FXS on the TDM400P. The problem

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

Hey Guys I have an indial range of 61211118[01234]X being trunked sip to xxx.yyy.189.65 Now I want to break this down into 612111180x going to xxx.yyy.188.145 and 612111184x going to xxx.yyy.189.199 reminder being used for fax->email etc etc etc I have created the outbound routes and sip trunks I can see that all the sip trunks are up I can see the outbound routes are there and

Does anyone have any suggestions for connecting analog DID trunks? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4

We have working in a rural project and I have selected Asterisk. Because * will work with a prepay system we need a precise answer supervision for outgoing calls. The pilot system will have 8 trunks to PSTN. The Telephone Company provides EURO BRI and analog CO trunks. On CO trunks the answer supervision is reversal polarity. The village to be served will be connected through a WLAN. Thus,

I have asterisk with two zap channels which are analog ports off a T1. They each have a inward DID number If they are used for outgoing they show the T1 main number not the DID's number. Is there any way to send caller ID of the inward DID number not the main number Jeff

I am trying to turn up DID trunks with our local phone company but do not know the correct format of extensions.conf to do this.

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Hi, I manage an SBC which stands between my company server farm and some SIP telco trunks. The system works fine, for inbound and outbound calls. Now I've configured the SBC to also act as a registration proxy, forwarding SIP registrations coming from the Internet to my asterisk servers. It all seems fine, but it doesn't work well, because by the time at least one phone registers through

Hi guys, I'm new to the telco game and still pretty new to Asterisk, although I've been using it for a couple of months now and like most of what I see. At my office, we've got a small two extension setup w/ two Digium cards for a single FXO line and three FXS extensions, but I'm also working on designing a larger installation for a customer which will involve ~16 analog

Hi all, Im trying to build an small home system. I have 2 pots lines, and i need to make 8 extensions and be able to use my old analog phones. What would you recommend to use as the 8 FXS switch? I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS ports. But i don't know if it is the best solution. Does anyone have a better solution to build this system? If an analog