similar to: Conference bridge dimensioning

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 it... I buy a half a million dollars worth of Trashiba's finest ... I download Asterisk for free... I now refer to it as legacy 18 months and 300 extensions later! Anyway, I am trying to integrate my dial plans acrossed platforms. PSTN>>>CTX670>>>Asterisk The dialplan I would like to setup, 1xx,2xx,3xx,7xxx to the CTX 670 4xx,6xxx to a remote ctx100 (this is setup

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

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 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 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 three polycom 501 that are all hearing echo. The other party sounds fine but you can hear yourself rather well. The volume does help if lowered but that also makes the other party extremely quiet. Is there any way to control the gain of the mic or stop the microphone from picking up so much from the handset. It only happens while you are on the handset. -------------- next part

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

On 6/4/07, Jordan Novak <jnovak@logisticshealth.com> wrote: > I am having a difficult time with the transition from agentcallback login... > Here are a few of the isssues, I am logging in using chan_ local > ie:local/8000 as the extension I'm not sure if this will solve any of your problems or not, but I've found it's often necessary to use the "/n" on the

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

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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 agree with these fella's, this is a piss poor way of fixing it. I only know of one call center that used static agents, mostly because they were sold a peice of crap and they had no idea how to use it the other way. I think you will find the majority of call centers are callback centers. This decision has taken Asterisk out of the realm of providing reasonable call center solutions. VIVA

I have two IAX trunked *, there are loud crackles and pops, they are dialing out a T-1 and are sip devices, it also drops words, any help or Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060516/dc68274f/attachment.htm

I am using rrmemory for my queues. I have noticed that the application will only distribute or dial one number at a time. Is there a different strategy that will allow the queue to distribute more than one call at a time? I don't want to use ringall because that would tie up thirteen of my trunks every time it tried to distribute a call. Any thoughts? -------------- next part -------------- An

I am having a difficult time with the transition from agentcallback login... Here are a few of the isssues, I am logging in using chan_ local ie:local/8000 as the extension Call Detail records no longer show agent/xxxx as the dstchannel show agents no longer shows the channels state show queues does not show the member Can anybody help? I have a ton of time invested in applications I developed

On Fri, Jul 17, 2020 at 11:57:57AM -0700, James Jones wrote: > Accept the DRM_FORMAT_MOD_NVIDIA_16BX2_BLOCK() > family of modifiers to handle broken userspace > Xorg modesetting and Mesa drivers. > > Tested with Xorg 1.20 modesetting driver, > weston at c46c70dac84a4b3030cd05b380f9f410536690fc, > gnome & KDE wayland desktops from Ubuntu 18.04, > and sway 1.5 > >

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

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