similar to: Canadian VoIP termination?

> The documentation mentions that the Digium channels can be split into some > voice channels and the remainder of the channels used for routing IP > traffic. > > Does any one have this in use in conjunction with Asterisk? Does it work > well? Would you recommend it for a production server? > > Obviously, if this works, this makes for a cost effective platform where

What types of ISDN BRI cards work well in Europe (Guadeloupe, Martinique and France) ? For example: AVM C2 or AVM C4 or eicon Diva server 4 BRI? Any others? Which driver is appropriate? Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -------------- next part

I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm a little unsure as to how get the phone to figure out which ip address it should register with when it boots. How do I do that? I already have a tftp server for my SIP based phones. Do I need a tftp server for skinny configs at all? And if so, can it be the same tftp server as the SIP ones use (I'm not sure

Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-us&page=byoa/plus&ST=1

All we have a deal on Canadian termination. Rate: $0.0039 US Dollars Billing: 1/1 Protocol: SIP or H323 Codec: G729 Terms: Prepaid Only. We have a real-time web interface where you can monitor or download your CDR's. Please e-mail me offlist if you are interested: jweisman@ibell.net Thanks, Jon -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for

I read about this intercom stuff on page 62 & 63 of the book "Developing Cisco IP Phone Services" isbn 1-58705-060-9. Primary calls take place on streaming channel 0. When streaming channel 0 is not in use, streaming channel 1 can be used for asynchronously streaming (in and out) stuff like voicemail, email, and, yep the one we want, intercom. Page 87-88 of the book talks about

I assume there are several people on this list that have Cisco Call Manager implementations under their belt.... We are beginning a call manager implementation and the first question I asked Cisco was, should we use SIP or Skinny. Cisco is pushing me towards Skinny, saying that I will lose some functionality with SIP. They also say that most of their customers implement skinny. I see two

We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. We have a flash frontend thats tied to our backend mysql DB. We use it for loading web site traffic data, email opens, click-throughs, bouncebacks, stats, etc. It could also be used with

i'm looking at potential asterisk service providers and came across oneunifed.net i googled for opinions and feedback, but haven't come across anything yet. is anyone using them or does anyone have feedback on their asterisk support and expertise? tia, george

Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040116/aa4eda3c/attachment.htm

hello, we have been using a couple of US based VoIP providers for outbound calls completed within the US, without any issues. We recently started making calls to Canada and have received a few complaints about the call quality. Questions : - Could this be because of the number of intermediate IP hops between us / our VoIP provider and the Canadian phone companies ? -

Hi, Does anyone have experience with doing Canadian cross-builds of LLVM compiler? I need some help here... I'm a new with LLVM, so please don't be too harsh on me. I am building LLVM cross-compiler (and as I mentioned I'm doing it via canadian cross-build): Build platform = Linux, Host platform = Windows Target platform = Linux I'm using mingw32 compiler toolchain to

Hi All Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are fine. David Kwok -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e8023f35/smime.bin

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

I need to get 2 Asterisk servers working together. I have been reading and doing just about every example I have been able to find here on the list and the Wiki. It's now gotten to the point that nothing on box2 seems to be working. I seem to have a major problem understanding the format. Here is what I have so far. It's 3 days of hair pulling and nothing seems to work! Asterisk box 1

My apologies for the cross-posting. If you are a business or individual providing Voice over IP services in Canada then we encourage you to read this email carefully otherwise please disregard. ----- As you are most likely aware, the CRTC has undertaken the roll of regulating VoIP services in Canada and is currently conducting hearings with the goal of putting in place regulatory requirements

Just on the off chance that Canadian Asterisk users might be interested in a place to discuss topics specific to the "great white north" (sources, services, telcos, etc.), I created the asterisk-canada mailing list: http://lists.syonex.com/mailman/listinfo/asterisk-canada or asterisk-canada-subscribe@lists.syonex.com Cheers! John

Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten => 1637,1,Dial(SIP/100) Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _NXXXXXXXXX,2,Congestion Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _1NXXXXXXXXX,2,Congestion These

Quoting asterisk-users-request@lists.digium.com: > > Would it be considered trolling to start a thread on Cleaning Maple > > Syrup off of Dial Pads, or Wiring your Moose for Wi-Fi? > > Let's not forget the weekly "tooques and telephony" segment, and a > review of > the best block heaters for your wi-fi fones. > Oh, we're gonna have a good time next