similar to: How to verify availability of the DID connection?

Can anybody provide DIDs for Chatham, ON? Usage based preferred, but flat-rate is not an issue. ? ? Contact off list. ? Thanks for your time, ? ? Sincerely, Robert Augustyn ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100529/e8ac734e/attachment.htm

Is it possible to have Calls incoming to different DIDs? I want an AA that handles 100s of businesses. [Incoming-pizza] Exten => 4045551212,1,Goto(pizza,s,1) [Incoming-hvac] Exten => 8085551212,1,Goto(hvac,s,1) [Incoming-gutter] Exten => 6175551212,1,Goto(gutter,s,1)

> > There is a big confusion here about Stir Shaken. It is NOT a provider > issue. Un fact, all providers are whasing their hands and modifying their > swihtches to pass-through the Signature. They cannot sign the call because > then the become the responsible party for the call before the FCC, and > liable for any illegal call. Every owner of a PBX that sends calls to the >

In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls in our servers to sign them. We do this over a MySQL call, easily connectable to Asterisk via

Hello, I am trying to piece together a solution to do the two things: 1. receive faxes on a bunch of numbers, about 10,000 2. receive voicemails from people on the same numbers At this point, I am just trying to see if what is required can be done. The faxes need to be received and put into PDF and shipped off to another server somehow. That can be FTP, UNC, NFS, something. This can be a

I am interested in setting up asterisk to record all calls it processes. there are however some legal quirks to doing this that I have run across such as one party notification vs 2 party notification requirements which depend upon the physical endpoints of a call. If I wish to discretely record all incoming calls except those calling from a place requiring 2 party notification (thus avoiding

Hi, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a strange problem, when I try to connect to les.net from our local asterisk server through Procurve router I seems to be connecting on any port above 1024 and when I reload sip the port is changing too ... So I never get 5060? Any ideas on what is going on and how to resolve it? ? Sincerely, Robert Augustyn 519-997-3106 ext:802 www.linqone.com ? ? -------------- next part

Hi, I am planning couple small business installations and wader what should I use for 2 to 6 lines a gateway or pci card? Any comments greatly appreciated on pros and cons and brands. Thanks, robert

Hi all, I'm quite new to R and have a very basic question regarding how one gets the standard error of the mean for factor levels under aov. I was able to get the factor level means using: summary(print(model.tables(rawfixtimedata.aov,"means"),digits=3)), where rawfixtimedata.aov is my aov model. It doesn't appear that there is an equivalent function to get the standard

Hey; I figured I would ask here as I seem to get better results. I am using the voip.ms <http://voip.ms/> VoIP service. I have no problem configuring my Asterisk server 1.8x to dial out with my Softphone. HOWEVER, for some reason, I cannot get inbound. All that I hear is a busy signal. I know this is not much for you folks to go on, but what would be a good place to start

Hi, We have zabbix running and would love to be able to monitor our asterisk box with it. I believe that some sort of SNMP is build in 1.4+ correct? Where do I find more info or a how to on what is supported and how to use it? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Is using a Citel gateway with Asterisk a good solution for reusing of the old Nortel digital phones? Would love to get some input from actual users. Any/all opinions welcome. robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070510/bc5fc18f/attachment.htm

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

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

there is one more thing that you should probably see: this is the error message that cygrunsrv.exe gave me: Eric at ballistic ~ $ cygrunsrv --start sshd cygrunsrv: Error starting a service: QueryServiceStatus: Win32 error 1062: The service has not been started. this is the error message that "net" gave to me: Eric at ballistic ~ $ net start sshd The CYGWIN sshd service is starting.

Hi! Samba 2.2.4, Linux. smbd loses connection to the PDC - although rest of organization feels fine... I've had the following cropping up: Jul 25 07:40:13 10.17.0.2 smbd[6994]: [2002/07/25 07:40:13, 0] lib/fault.c:fault_report(38) Jul 25 07:40:13 10.17.0.2 smbd[6994]: [2002/07/25 07:40:13, 0] lib/fault.c:fault_report(39) Jul 25 07:40:13 10.17.0.2 smbd[6994]: Please read the file

I know that Asterisk supports DID, but does anyone have documentation on how to write the configuration for it? I'll be trying to setup a hybrid system where some incoming numbers will be DID enabled and others won't, so I'll need to be able to sort between the two, i.e. directly connect the DID dialed numbers and route the others to an autoattendant for extension dialing.

Hi, all. I'm trying to do some rudimentary testing of an Asterisk system, but, for various reasons, I have to do this covertly, which means I'm paying out-of-pocket. So I'm looking for somewhere that will do *cheap* SIP and/or IAX termination, preferably with at least two simultaneous calls, and one DID. Any suggestions? Thanks, -Ken

I am setting up a box with a T100P. Everything is going well. The company I am working with has their one phone switch gear. They provisioned me a E&M Wink T1. Cannot do PRI unfortunately. We chose E&M so we could pass an unlimited number of DIDs to the trunk as apposed to FXS loopstart signaling. I can make outbound calls no problem, but I am having problems with the dial plan for inbound