similar to: Web based call control

I am using Asterisk to send Voicemail out as Email. I am running into a problem I believe to be caused by the exchange server requiring SMTP authentication. I cannot get the sys admin's to turn it off. Does anyone know enough about sendmail to help me. I am assuming that the default mail client is sendmail. It will also send to other non-SMTP authenticated servers. Your help is much

I have installed asterisk on numerous servers. Every install was done on Fedora and (White box Linux). I now have zap channels in one of the boxes (T-1). No matter what type of channel I call on (sip or zap) I get some really strange artifacts in the sound, almost like a skip in the playback. It seems to always be in about the same place in the recording. Usually in the beginning of playback. For

I love the added apps installed with trixbox, ARI, Web-Meetme, FOP, and Reports are great. FreePBX on the other hand, is nearly impossible to do everything with. Trying to edit the configs manually proves impossible due to the excessive use of includes and macros. It is kind of like watching someone try to bite their own ear off. Has anybody tried to wipe all the configs clean and program the

Can someone point me to instructions on how to install, I have edited the defines.conf and set up the database. I have apache running and have no clue what to do now. I have NO experience with php based stuff. HELP!!! Jordan Novak Communications Technician -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a question about how Asterisk Parses the Dial Plan. To create a hunt-group which would be the appropriate dial plan: [CompanyABC] exten => 7228888,1,Dial(SIP/8017228888,60,r) exten => 7228888,102,Dial(SIP/8014361234,60,r) exten => 7228888,103,Dial(SIP/8014362345,60,r) exten => 7228888,104,Dial(SIP/8014363456,60,r) exten => 7228888,105,Dial(SIP/8014364567,60,r) exten

This has to be the worst documentation I have ever come acrossed. I have found two or three docs on how to install it, but they are all so different and make huge assumption about what packages you have installed and locations of files. Has anyone seen something better, I want to get this working it is quite a cool app. Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint

I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has

I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so... I had to modify the 01-devfs.rules Make linux26 Make Make install... Everything appears to compile correctly but it says module not found when doing "modprobe zaptel" Is this a rights issue? Jordan Novak -------------- next part -------------- An HTML attachment was

I have a box that will send to my personal pop/web based email but will not send to my exchange server. I have checked the MX record and DNS settings. I know there is something you can do like this to check it but it returns either a -1 or 0 (have no idea what that means) sendmail /mx anyway I can send to a ISP based Mail account outside the company. We have .wav files allowed we also require

Is there an app or softphone for meetings that displays the hosts screen like webex or intercall. Jordan Novak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060321/df90d527/attachment.htm

I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/88e22671/attachment.htm

We are using an Asterisk box to do conferencing right now. I have had about sixteen active lines in conference and the quality was acceptable. We now have a need for 50 people to conference at one time. Does anyone have enough experience doing this to give me some pointers. Will it even be reasonable to try this? Is the mixing done on the the hardware, I plan on using a quad span t-1 card from

I am aware that there have been several posts related to ssh connections hanging, i have tried to read through most of them in the archives... I am posting this in hopes that it may present something new that will further help resolve this problem. The problem i am having appears to be similar to what others have reported where after typing exit in an interactive shell the connection is not

I am looking at some directions on how to install and it is asking me to edit defines.php, it states that the file should be located in the source directory, but I can't seem to find it anywhere on my machine. Anyone been thru this? Jordan Novak Communications Technician -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am not getting any caller Id with my standard T-1. Is a standard "T" capable of sending callerid? I don't want to spend time troubleshooting my PBX if Asterisk can't send it down that type of trunk. Jordan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am wondering how many and how often manager connections can be setup and torn down reasonably. here is the scenerio... I have 10 to 20 agents on two queues one with priority over the other I changed this the day before I also implemented a php program that runs every 8 seconds on an automatic refresh It establishes a connection to asterisk and runs a mysql query to update the database

Hi, I am using wine-0.9.30 (self compiled) on a Debian 3.1 server. Wine is required for CUI-applications, so no X-Server is running. When I log in via ssh and start the application, wine is working fine. But whenever I try it from the webserver via php-script, wine doesn't start properly. It fills the errout with "epoll_ctl: Operation not permitted" unitl I send a SIGKILL to the

How to define a MachineFunctionPass requireing one of the register allocation passes being executed first? Should there be a PassInfo for register allocation in Pass.h? Pass.h: namespace llvm { extern const PassInfo *PHIEliminationID; extern const PassInfo *TwoAddressInstructionPassID; extern const PassInfo *RegisterAllocationPassID; // <-- add this one? There are four

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¢

I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a "core restart now" cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2