search for: packham

...spent a lot of time on making it user friendly…. Just working good and parsing the correct stuff out. I hope to release the code to Mark soon. Any comments welcome, just look for me on IRC (p0lar) or use the list. Here is the link http://warpcore.netcom.utah.edu/openconf/openconf.php Dave Packham University of Utah NetCom Manager R & D D.S.O. University Incident Response Team c. 712-0364@801 w. 587.0907@801 Dave.Packham@Utah.edu Trillian/ICQ#:45818442 Current Trillian status: 45818442@pager.icq.com SMS: (Send an SMS message to my ICQ): +278314245818442 More ways to contact me:...

I have put my phpconfig stuff out into the Digium CVS tree. Project name is phpconfig. see it at http://rads.netcom.utah.edu/phpconfig/phpconfig.php Lemme know if you have any patches or add on's are welcome Dave Packham aka p0lar

has anyone done caller announce in MeetMe's before? Dave P >>> brian@bkw.org 5/18/2004 5:50:49 PM >>> With multiple parking lots you can give each person their own lot thus exten 800 for everyone will connect them with just their call passing the lot name which you know for X customer. bkw ----- Original Message ----- From: "Andrew Kohlsmith"

I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the

does anyone have a recorder GSM file that emulates the Telco's "if you are a telemarketer please hangup now" recording? I don't see one in the sounds dir. the ZapATEller works great for computerized callers but if a human hears this message asking them to go away they have to. Isn't that right? Dave

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

...P/70907|30 exten => 70907,2,VoiceMail,u70907 exten => 70907,102,VoiceMail,b70907 exten => 70907,103,Hangup {70902} ; Andrew's Extension exten => 70902,1,Dial,SIP/70902|30 exten => 70902,2,VoiceMail,u70902 exten => 70902,102,VoiceMail,b70902 exten => 70902,103,Hangup Dave Packham

Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _________________________________________________________________ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now!

...inejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial. would the sip debug help? ill post if you need Thanks Dave Packham I have this in my extensions.conf and have tried both of the below options TRUNK=Phone/phone0 ;[trunkld] ; ; Long dist...

...rification on the manager.conf as well as read/write features based on the manager.conf... I am getting it ready to give to Markster to include (if he wishes) into the cvs tree. I would accept any constructive/positive as well as well thought out slightly negative comments and diffs... :) Dave Packham U of Utah >>> steve@LawNet.Com 6/26/2003 7:04:59 PM >>> Well, for *, I fall into the newbie category (not for telephony, VOIP, Internet, *NIX, C, etc - those I've been doing since the Internet had 3 nodes :-) and each technology mentioned was a newborn) I believe making i...

Before you get too far.... The internet line jacks dont allow outbound calling. they cannot be used as trunk lines to the PSTN. the outbound code has not been written yet. I had to go buy FXO card from digium (that works much better than the Linejack) to get outbound calling to work Dave >>> fplandae@hotmail.com 4/11/2003 5:35:24 PM >>> Hi, I am a newbie. I have been

http://www.nero.com/us/631911127302064.html Have you all seen this? Its a SIP softphone put out by the people that do the CD burning software Nero... Check it out it works with * Dave

Hi, Not sure if anyone will be intersted but I have put together an install guide for Asterisk on RedHat 9.. Its nothing special but it may be of use to the newbies.. Like me.. If you would like a copy let me know and I will send it to you.. later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

...p://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; <?PHP /** * * Asterisk configuration file interface script * * * * * * phpconfig:,v 1.0 2003/07/03 17:19:37 * Authors: Dave Packham <dave.packham@utah.edu> * Rob Birkinshaw <robert.birkinshaw@utah.edu> */ require_once("phpconfig_init.php"); require_once("cls_phpconfig.php"); require_once("cls_phpconfig_html.php"); // create and initalize objects // $page...

...p://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; <?PHP /** * * Asterisk configuration file interface script * * * * * * phpconfig:,v 1.0 2003/07/03 17:19:37 * Authors: Dave Packham <dave.packham@utah.edu> * Rob Birkinshaw <robert.birkinshaw@utah.edu> */ require_once("phpconfig_init.php"); require_once("cls_phpconfig.php"); require_once("cls_phpconfig_html.php"); // create and initalize objects // $page...

say I needed to send a broadcast message that I wanted every user to hear when the pick up thier phone? can I "Play,message" on a line just before they get dialtone? or maybe after they dial before ring? how about a "ringdown" to a voicemail box and on end return them to thier line for the dialout? can * do ringdowns? when a user picks up an extension it automagically

I thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

...oad the page. > See some of the output from http://ip-of-machine/phpconfig/phpconfig.php > below; > > <?PHP > /** > * > * Asterisk configuration file interface script > * > * > * > * > * > * phpconfig:,v 1.0 2003/07/03 17:19:37 > * Authors: Dave Packham <dave.packham@utah.edu> > * Rob Birkinshaw <robert.birkinshaw@utah.edu> > */ > require_once("phpconfig_init.php"); > require_once("cls_phpconfig.php"); > require_once("cls_phpconfig_html.php"); > > // create a...

...rting my search for a SIP/MGCP/H323 channel bank. I need analog ports in a building that only have network connections back to my * server. I could install another * server and use a normal CB with a PRI but I would like to investigate any good CB's with network trunk abilities Thanks Dave Packham

*This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, I've just done a quick (but functional) web front end for searching the CDRs in a MySQL database. Anyone interested in trying it out? I'm wondering what to add to it next. So far you can seach using source, destination, CLI, channel and date ranges. It also displays ALL fields in the database table.