similar to: SIP vs RTP destination IP

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

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'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'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 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

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

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 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've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure what it's supposed to do, but I wouldn't expect it to continue processing the dial plan. Any pointers? Documentation locations that address hanging up would greatly appreciated! TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david@safedatausa.com

I'm going to need to build a few Asterisk boxes that have dual and quad T1 interfaces. I knew Digium made T1 interface cards and on this list I heard about Sangoma so I did a quick search and found the hardware page at voip-info.org which lists several manufactures I didn't know about. All that leads to this question: I'll be using T1s in the USA. What experiences have you all had

(missed one thing) I have made some changed to your config: extensions.conf from 192.168.253.21: ; ; Create an extension, 205, for trunk ; exten => 205,1,Dial(IAX2/ <mailto:IAX2/192.168.253.20/205@iax-trunk> tecinfo1/205) iax.conf from 192.168.253.21: [tecinfo1] type=peer username=tecinfo2 secret=secret host=192.168.253.20

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 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

I'm trying to use the password entered with Authenticate to create dynamic meetme conferences with the following dial plan: exten => _XXXXXXXXXX18467,1,Authenticate(/etc/asterisk/meetme.pw|a) exten => _XXXXXXXXXX18467,n,MeetMe(CDR(accountcode)) ; 281-8467 However CDR(accountcode) is always being set to 1022 no matter what password is used. The passwords are stored in a file so they can

I just got a brand new A101 and am trying to install it in my test Asterisk box. The install went without a hitch. I followed the directions on the Sangoma Wiki: Wanpipe Asterisk Install http://sangoma.editme.com/wanpipe-linux-asterisk-install Wanpipe for Asterisk Configuration http://sangoma.editme.com/wanpipe-asterisk-configure It went perfect, no problems and Asterisk came up fine. I downed

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