similar to: Mandrake zaptel module not found after compiling

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

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

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

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Hi, I am investigating getting a wifi VoIP phone because its may be a better option than an ATA and a cordless phone.. Does anyone have any experience with the whats out there?? Do they support things like WPA etc?? I have heard the battery life can be a problem.. Is this the case? Thanks..

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

Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. Thanks, James

Has anyone got this annoying sidecar to illuminate when users are on the phone? In my function key settings I have: Context: Active Type: Extension Number: <sip:4000@serve.address.com;user=phone> (4000 is the extension I want to see/dial on the key). I can press the key and it will dial the extension, it just won't illuminate when the user is on the phone or on DND Since I have

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:

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All

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 have a client that has 5 Snom 320s. 4 work great, one does not. I upgrade the firmware to the latest (6.5.2) and the problem goes away, but then comes back a couple days later. There is a slight packet loss on the phone (about 1%), though there is no packet loss on any of the other phones. I determine the packet loss by the Linux command "ping -f -c 10000 192.168.2.10". Outgoing

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

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

Does anyone have a guesstimate of how many active Asterisk installations there are? Sorry this is off topic, need it for a customer proposal and they need comfort on stability. A count of the downloads from Digium would be a good start but I couldn't find this anywhere with Google. Feedback from anyone who may know near actual data would be appreciated rather than simply guessing,

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

Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display "Confidential" or "unknown" as we sometimes see ? Andre

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

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