similar to: Queue application strategy

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

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

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

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 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 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

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 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 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

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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

> -----Original Message----- > From: Chris Murphy > Sent: Tuesday, February 17, 2015 20:48 > > On Tue, Feb 17, 2015 at 7:54 AM, Jason Pyeron wrote: > >> I'd post the entire dmesg somewhere > > > > http://client.pdinc.us/panic-341e97c30b5a4cb774942bae32d3f163.log > > At least part of the problem happens before this log starts. Feb 15 23:41:19

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 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

> -----Original Message----- > From: Chris Murphy > Sent: Tuesday, February 17, 2015 3:58 > > I think the panic is the consequence of drive write failure. > So the actual > problem is before the panic call trace. Most of the time it panics without any warning, but once there was: > > -----Original Message----- > > From: Jason Pyeron > > Sent: Sunday,

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

We're using gluster-2.0.1 and a nufa volume comprised of thirteen subvolumes across thirteen hosts. We've found today that there are some files in the local filesystem associated with the subvolume from one of the hosts that are not being seen in the nufa volume on any gluster client. I don't know how or when this happened, but now we have to do some work to get this gluster volume

NOTE: this is happening on Centos 6 x86_64, 2.6.32-504.3.3.el6.x86_64 not Centos 5 Dell PowerEdge 2970, Seagate SATA drive, non-raid. I have this server which has been dying randomly, with no logs. I had a tail -f over ssh for a week, when this just happened. Feb 8 00:10:21 thirteen-230 kernel: mptscsih: ioc0: attempting task abort! (sc=ffff880057a0a080) Feb 8 00:10:21 thirteen-230 kernel:

Hello, I am synthesising some sales data over a twelve month period, and then trying to use the "predict" function, firstly to generate a thirteenth month forecast with upper and lower 95% confidence limits. So far so good But what I then want to do is add the upper sales value at the 95th confidence limit to the vector of thirteen months and their respective sales to create a