similar to: Setting DID info for analog Zap channels

Well this is disappointing. Time to find somebody else... -- Wes -----Original Message----- From: NuFone Operations [mailto:support@nufone.net] Sent: Tuesday, April 18, 2006 3:44 PM To: wbaehr@totalmac.net Subject: NuFone Update: DIDs Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier supporting the Toll-Free and Michigan DID operations of NuFone, has threatened to terminate

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

I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read "Asterisk: The Future of Telephony" and done a fair amount of googling. I am completely sold on Asterisk, and the new building's

Hello, I just wiped out my old asterisk install and installed Asterisk at Home. I was quickly able to get my Digium TDM422P working, 2 POTS lines, 2 phones. I also got X-Lite working as a SIP extension. I then tried to setup my Polycom IP 500, and this was not so easy... Using AMP I created SIP extension 205 to be used with my Polycom phone. I setup username = 205, secret = 123, context =

Hi everyone: I've searching for a while and haven't found what i need. The thing is that i have a tdm422p with the two fxo ports connected to the pstn. I want my sip users to be able to call other numbers(any number) in the pstn through my zap fxo channels. I have a big number of sip users so as you can imagine there will be congestion when some of them(more than two!!) want to call

I have server with a total of 6 Analog ports...using TDM04B and TDM02B. I have 3 Lines that are DIDs and 3 are Main/Roll Over lines and I have worked through getting the DIDs to work and route to the extensions...now what I need to do is when Extension 1111 picks up the phone to dial, I would like them to use their DID analog line first, unless someone has called in on it and they are trying to

Hi all, This is not a repeated post as I am just adding more information for my previous post. Asterisk version 1.4.18 TDM card: Digium TDM411B Zaptel version 1.4.9.2 Line: PSTN line I tried to obtain the dialed number using $DNID and $CDR(DST) . All of these variable returns 's' I also tried exten => _3345335,n,Noop(this is ok) where 3345335 is my number but it does not go there.

Hello, our gateway receive incoming calls from an outside gateway for multiple DIDs. For some of them we want fax detection, for other no. To do so, faxdetect is set to yes, but how to disable the fax detection for a specific incoming DID? For those DIDs, we just want to forward the call to a real fax machine DID which will do the job. Thanks for any hint Regards -- Daniel

Hi all, Occasionally, DIDs from different providers stop working for some reason. I would like to be able to monitor situations like that and react before any of my clients start going ballistic on me. Any ideas? Scripts you know of or wrote and willing to share? Any info?would be greatly appreciated. ? Robert ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I am experiencing some intermittent problems with calls coming inbound on my DID trunk. I have 12 DIDs that come into an Adtran 750. From there T-1 to a port on T400P. The problem is that some calls that come in don't seem to bridge properly. Heres what happens. Call comes in on Trunk. Call Routed to correct Zap Channel. Phone Rings. Person Answers phone, but hears nothing but

I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like

Cross posted from asterisk-biz: > > Is anyone willing to host/manage a website that people > > can simply browse that lists all current DID providers > > and their coverage areas? > It's a good idea and probably not too hard to implement, > it's just a case of deciding how far you want to go.. are > areacodes good enough? or do you need to go to NPA-NXX >

Hello, I'm looking for US DID and US50/CA 800# Providers. I found voiceconduits.com 8 month ago, there interface looks good, but there are still not live, I believe they won't be any time soon. I found sixtel, but order take eternities, they probably won't get my orders right any soon. So i'm looking for a good provider for DIDs and 800# from the US and CA, who offer online

I have been looking at moving from SIP-based DID (Illinois) providers to one that uses the IAX protocol for DIDs. After a search, I've come up with the following: http://connect.voicepulse.com -- $8/month, many rate-centers http://www.iax.cc -- $1.50/month + 0.014/min, many rate-centers Can that be all that there is? I like the pricing plan at iax.cc, because it would allow me to set up

I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the

I want the ability to setup DIDs in a variety of different remote locations in Canada. There are various providers that have DIDs in major cities, but none that focus on the "smaller" cities. The question is how do I actually setup these DIDs? Thanks, Geoff

Hi all, Aside from the obvious answer "write your own", anyone know of any DID billing software out there? I'm looking to possibly resell som DIDs to people and want to charge some per/minute rate when forwarding to their ATA devices. I can pull the data off mysql and create invoices but I'm kinda looking for an app similar to ASTCC, A2Billing, etc. where everything happens

Anyone recommend anyone who can provide me (actually a customer, but I'm asking on their behalf) DIDs in Sweden? They already have an asterisk box (in Sweden), now want a local number for it! Thanks, Gordon

I've got a run of Shielded Twisted Pair (4 conductors) which used to be a Token Ring Network drop and I'm not using it anymore. Would it be decent to replace the ends with normal analog phone connectors and use it for a phone extension, or is STP unsuitable for that? Thanks Steve

Hi - I am looking for suggestions for ITSPs for faxing with asterisk 1.8. We are based in the US, so would need an ITSP with US DIDs. #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38