similar to: 'Stop Now', 'Restart' problems

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

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

> 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

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

Hi, I am trying to get the extension status (weather it has dialed outgoing call via SIP or IAX2), using the following piece of code however it always returns -1 on all the extensions (valid/invalid). Am i missing something ? Any help. Thanks ----------------------------------- #!/usr/bin/perl use Asterisk::Manager; use lib './lib', '../lib'; $|++; my $astman = new

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

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'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

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Do the Polycom IP phones have some programmability so you can do some programmable phone buttons like you can on the Cisco phones? If there is programmability, such as for soft-keys and the like, how would you rate Polycom's vs Cisco's capabilities? And where can one find the programming documentation? Thanx. Ray. -- Scanned for viruses and dangerous content at

Hi A small question on Asterisk Manager. I use Perl Script for start a call: my $response = $astman->sendcommand( Action => 'Originate', Channel => 'SIP/ASTERISK/$Extension', Exten => '200', Context => 'MyContext',

>I spoke with someone today who is interested in an IP Centrex solution that >starts with about 3500 extensions in a multi-tenant application. And >growing from there. > >I'm wondering about scalability of Asterisk. I'm trying to put my head >around how to put the whole thing together, if it can be put together. > >The nice thing about it is that if I can show

Cisco has a part number SW-SM-UL-7960 for licensing SIP on their CP-7960 phones. Is this actually required to be purchased to keep everything on the above-board when using Cisco's SIP phone with Asterisk, or is this for something else? Ray. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean.

Any one see these? Are they benign, or is some system tuning required to remove them? Can't seem to find a resolution in the archives. If you have a link, it would be appreciated. Jun 2 10:58:58 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 19 received Jun 2 10:58:59 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 72 received Jun 2 10:59:00 NOTICE[163044272]:

Anyone connected Asterisk to hospitality packages such as: Micros Fidelio Visual One Jonas We'd be interested in providing bounty on providing a connection to one or more (depending upon what the client selects) if our proposal goes through. Ultimately, about 300 to 600 stations will be provided. Ray. ------------------------------------------------- This mail sent through IMP:

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

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, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...