similar to: Call load balancing

I've got a couple of PRIs. When I call out on them from internal SIP phones, I will get ringing if the dialed number is ringing, but if the dialed number is busy I'll get dead air. Can anyone suggest ways to trouble shoot this? Don't seem to having any other problems with the PRIs. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at

I'm working on writing some test IVR code in AGI. I can't get my FXO port to detect a hang-up, but I'm going to deploying this using Digital cards so I decided to just skip that problem for now. However this leaves me with a question. How does AGI detect a hang-up if everything is operating normally? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc.

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

I have written a script that is executed using ExternalIVR(). I am running in to performance issues when I have four or more simultaneous calls running this script. I'm running on a P4 2.8 with 512M, all calls are GSM coming in over IAX from an asterisk box that acts as a switch and handles all PSTN interfaces. My question are these: Are there ways of optimizing ExternalIVRs? (maybe

Hi, How do you folks handle really large queues (350+ simultaneous callers) in your Asterisk PBXes? We're going to be bringing in around 16 PRIs' worth of inbound callers, doing skills-based routing, and queuing them up for approximately 200 agents. What's the best way to handle all of these callers? We want to record the calls and we'll probably use the ramdisk method that has

I installed a Sangoma card with the default install. I'm getting five digits of DNIS with each call. The T1 is setup ESF/B8ZS wink start. Each of the digits of the DNIS are being used for extensions in the context. I need a single extension that let me start an AGI script that can use the dnis. Can anyone point me in the right direction to do this? Thanks, David Ruggles CCNA MCSE (NT) CNA

Need some help with IAX trunking. I've got six systems: AsteriskM (main) ___________________|____________________ | | | | | Asterisk1 Asterisk2 Asterisk3 Asterisk4 Asterisk5 AsteriskM has two Sangoma A102 2 Port T1 cards in it, the other Asterisk boxes are using ztdummy for timing, they are all using IAX trunking. My calls come in

I've opened a few bugs on ExternalIVR and added patches. The biggest issue is: https://issues.asterisk.org/view.php?id=16174 [patch] ExternalIVR does not handle arguments in a consistant manner Basically, this optimizes and fixes several different ways of calling ExternalIVR. If there is anyone who is using ExternalIVR today and/or is willing to test these patches I would appreciate the

I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the

The subject pretty much says it all. Does Asterisk support DNIS, and if so, what kind of connection is required? (T1, PRI) I've got a wink start T1. I've read comments that say the DNIS will be seen as an extension, but I'm seeing each digit of the DNIS as a separate extension. So in my case I send DNIS of 12345, Asterisk will jump from extension 1 to extension 2 to extension 3 to

I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

I'm afraid I already know the answer because I've done a lot of searching, but does anyone know of a softphone that supports a central phone book and paging (like the sip autoanswer option of some hardphones) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

I did some simple AGI programming several months ago. I have a need to use one of those old programs and I'm having a stupid problem. I can't get output to display on the console. I'm sending it to stderr and I've got verbosity set to 10. I know I had it working before so I'm guessing I just forgot some piece of key information. Any suggestions? Thanks, David Ruggles CCNA

I'm reading voip-info.org http://www.voip-info.org/wiki-Asterisk+cdr+mysql Sorry if this is a dumb question, but: It says I need mysql and mysql-devel to compile cdr_mysql, but I don't want mysql on my asterisk box I want to connect to a remote mysql server. Can I use mysqlclient and mysqlclient-devel? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910)

I finally got my X100P working and now I have a question. I have several Skutch phone line simulators. My X100P works as expected with both a POTS line and an analog PBX port, but when I use a phone line simulator it doesn't answer the line. The phone line simulator doesn't power the line until the phone set goes offhook. Asterisk shows the RED alarm and then the alarm clearing but never

Is there any way to use Festival from script called by the ExternalIVR() dialplan function? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david@safedatausa.com

I've got a weird problem: I've added a new phone and "sip show peers" shows a status of "OK (x ms)" but when I dial it I get "status is 'UNKNOWN'" Any help on how to troubleshoot this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

First, I didn't realize I hijacked another thread! Please accept my apologies. Now the problem: Asterisk isn't compiling chan_zap. chan_zap also doesn't appear in the list of channels when you "make menuconfig" I have read all the replies and specifically Cosmin's and Tzafrir's emails. zaptel.h is located in /usr/include/zaptel I also tried "./configure

I've done some googling and searched voip-info but I'm not able to find a good answer about how to provision the GXP 2000. Based on questions I've asked before it seems like a lot of people are using the grandstream phones so I figure provisioning can't be that hard. Is everyone using the web interface for *every* phone? Or is there a better, more automatic, way? TIA!!! Thanks,

I have a bunch of sounds that I've converted into gsm from (Indexed NMS) vox files. There's only a single utility that I've found that can read and convert vox files. My conversion process is to use this utility to convert the index vox file in to a series of wave files and then use sox to convert the wave files to gsm files. Over all this works really well, the problem is that about