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I apologize about doubling these up, I forgot the subject! I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this... Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal

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 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

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

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 using Mitel 52xx dual mode phones in SIP mode. They work excellent, I am however having a problem with Voicemail retrieval. The Mitel Phones have a voicemail button on them. The light lites and clears correctly but I am not able to retrieve the voicemails using this button. In the phones web GUI it has a field for a voicemail server and port under additional servers in the networking tab.

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 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 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

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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

We are attempting to configure SIP trunking between asterisk 1.2.22 and a Mitel 3300 CX box. The Mitel machine will gateway to the PSTN for us. I found this earlier post about doing this from July: http://lists.digium.com/pipermail/asterisk-users/2007-July/191630.html Unfortunately the promised configs never came ;(. We're having the exact reverse problem: we can register with the Mitel

I have two issues... First I am working with a hotel software vendor to include an automated way to turn vmail on and off while clearing it at the same time. The vendor is looking to interface via serial cable as they currently do with Mitel systems. i am willling to work with them on an IP interface but I am not so sure on how to implement it in asterisk. Does anyone know of a way that may be

My Setup looks like this: Mitel 200 SX (1st T1) -------- Bell South (2nd T1) | | | Digium TE110P Asterisk MITEL CONFIGURATION Primary Timing Source: 1st T1 Card Secondary Timing Source: 2nd T1 Card ASTERISK CONFIGURATION span=1,1,0,d4,ami (Look to the Span for timing) We are getting a lot of Frame and Slip errors.... Time Slip Frame 7:00 736 950 8:00 690 1200 9:00 437