similar to: choppy recorded sounds in asterisk

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 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 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 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 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 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

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

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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

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

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

This is driving me crazy, when making an outgoing call the first 30 seconds is always perfect, then the party on the receiving end can always hear me perfectly but after 30-60 seconds the audio coming back to me from them starts to get choppy and drops out. I've tried this with multiple devices, from multiple locations some behind NAT, others not. This is using the ulaw codec, although