similar to: Identifying the Originating Cisco SIP Gateway

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

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

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

> 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

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

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

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

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

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

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

I'm not really looking for working configurations as much as I am looking for people who can say "This is a solid product and I trust my business to a solution running Asterisk." As far as pre-sales work... Well, tell that to my consultant. I'm quite excited about *. I've got my company sold on it, they just want some reassurance that it's ready for prime-time

If you're using perl on RedHat 9 make sure you put this command somewhere in your boot scheme: export LANG=C or at least execute it before running perl scripts. Redhat has EVERYTHING set to LANG=UTF-8 and it screws up all sorts of perl stuff, and several other pre-written programs in other languages too MATT--- -----Original Message----- From: Ray Burkholder

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

Hi, Does anyone know of Canadian VoIP termination providers? I have Canadian customers and would like to provide Canadian dial in and dial out (canadian callerid). Thanks!

Jeremy, In some posting in the mailing lists, you mentioned that docs for h323 had been submitted but hadn't made it into distribution. Could you post those docs in your download directory? I'm trying to understand the nuances of your driver, gnugk, and extensions. Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content

Ok, I've managed to get inbound and outbound calling to work with chan_h323 and gnugk. A few questions: 1) if I do a reload in *, chan_h323 loses its registration with gnugk, and will no longer pass calls to it. A second reload will crash *. Is this supposed to be? 2) For a configuration in h323.conf like: [office] type=h323 prefix=9 context=outbound I get a message saying:

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

Okay, I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our "traditional" PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. It is *not* an option to purchase a VoIP system package from Cisco, 3com, etc. Installers are getting an

I'm trying to come up with an elegant solution to handle roaming users in a branch office scenario. I have a number of possible scenarios, none of which seem to completely solve the problem. Perhaps someone with a better feel of the interactions can help me out. Is the 'switch' statement useful in some way? What are the ins and outs of the 'switch' statement? Come to think

Hi, I am having some problems with gigFit and would like confirmation on other platforms; mine is mint; basically Debian. Although I got a good fit for the density function with the GIG equation in another curve fitting program, I would really like to use R's tools for confidence intervals and manipulations; but the problems below make me uneasy. Problem one (from examples with parameter