search for: kurtwp

I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via DSL through a Linksys router. You can see the my sip.conf for the phone blow. The problem is each time the phone rings I can hear/be heard 50% of the time. Any suggestion on what to look for. I do have my reg time set for 180 seconds on the

>>> kurtwp@yahoo.com 6/15/2004 7:29:33 AM >>> >Old managers will change its the LaLawyershat don't >change. Every dam law office that I been in has at >least one fax machine that is constantly printing >something out. But to say fax is dead is an >understatement. >AT&T sai...

I noticed when I call certain IVR systems, such as 1800calldhl, that Asterisk will not barge the prompt. Would this imply that Asterisk has an Early media detection problem. Is anyone else experiencing this problem. Is there a fix? Kurt

I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by root@asterick.dell.cpu.com on a i686 running Linux Box B is running: Asterisk 1.0.3 built by root@dell.cpu.net on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12

Upon setting up and configuring the my extension.conf, meetme.conf and following the instruction outlined at this web page: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy I get the following errors when calling the meetme number. Executing Wait("SIP/216.53.118.2-f41196e0", "1") in new stack -- Executing MeetMe("SIP/216.53.118.2-f41196e0",

...emory leak in asterisk CVS (Erik Espinoza) 23. Re: SoftPhones: Bad, or just bad QoS? (Time Bandit) 24. Re: long pause on dialing.. (Randy Williams) ---------------------------------------------------------------------- Message: 1 Date: Mon, 18 Jul 2005 11:29:23 -0400 From: Kurt Pasewaldt <kurtwp@gmail.com> Subject: [Asterisk-Users] Asterisk Comedian Web page login To: Asterisk <asterisk-users@lists.digium.com> Message-ID: <723ac8b605071808296d5c212a@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 When I try to login into voicemail through the web interface It s...

I have two ATA186 running 2.14 and 2.15. I see in the SIP debugs that both ATAs keep on sending SIP registration packets over and over. The flow is as follows: Asterisk receives REGISTER packet Asterisk sends 100 trying, 200 ok with an expire of 3600. Kurt __________________________________ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time.

I am running Asterisk CVS-HEAD-07/14/04-16:28:29 and noticed that when I send a subscribe I get back a 403. This used to work in an old version which I forgot to record before upgrading to the above version. Any suggestion? I can register fine with the * server. Sip read: SUBSCRIBE sip:2486@192.168.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK46F2668 From:

Yes that is correct but most likely you wont't need/use every option in an Enterprise load. Also, an enterprise load will require more memory and flash; which which will increase the cost of a router. Kurt

I am trying to use the Directory() but am having difficulty using it. According to Wiki page that I found you need to pass it your voicemail.conf context. My vm-context is [local]. So when I setup the cmd (Directory(local)) I can search on the three letters of the last name find that user. But once I press one to except the name and dial the extension I get the following message form the *

I have the below script that works but for one problem. The call cannot last longer then 4 minutes when the script is utilized. However, when I configure my extension.conf to not call the script the call will stay up until I hang-up. I call the script as follows: exten => _24XX,1,AGI(internal.agi|${EXTEN}) exten => _24XX,2,hangup A brief description of the script is that it allows my

A user has their unavailable message played and once that message is over the Comedian message is played right after. Is there any way to prevent the Comedian message being played if the user's unavailable/busy message is being played. Thanks, Kurt

I would like to know if any body is using the Polycom Soundstation IP 4000 SIP conference phone with Asterisk. I am thinking of purchasing one. Kurt

I am having trouble locating webvmail on my * server. Is this a seprate porgram or does it come with *. I am running version asterick*CLI> show version Asterisk CVS-03/26/04-17:08:20 built by root@localhost.localdomain on a i686 running Linux asterick*CLI> Thanks Kurt __________________________________ Do you Yahoo!? Yahoo! Photos: High-quality 4x6 digital prints for 25¢

What I am trying to do is the following: A call is sent to the * box via a SIP invite. The * box answers via an IVR menu system with " Enter the extension you want to dial" so I enter in my 5 digit extension and get the below message. Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_request: No channel type registered for 'SIP)' Jan 18 10:10:03 NOTICE[-1380238416]:

Old managers will change its the LaLawyershat don't change. Every dam law office that I been in has at least one fax machine that is constantly printing something out. But to say fax is dead is an understatement. AT&T said that about teletype service, you know 50 - 300 baud service, years ago and then one day they noticed that traffic across their teletype seservicetarted growing. If

Folks, My question concerns the SIP Notify that is being sent to ... device. You can see it in the following line: Voicemail: 0/0 Shows no Voice mail but I did leave a voice mail at the extension. Any suggestion on what I should look for in my * setup. I am not worried about the 481 coming back for the other side yet. Once I get a handle on the Notify, I'll work on the 481.