search for: bdingman

I have been looking around for Outlook Integration for Asterisk. Saw the Asterisk TAPI wiki page and also ran across this: http://www.fonality.com/pop.cgi?page=pop_pbxtray.tt (PBXtray) It looks like Fonality has managed to make an app that does screen pops and allows click to dial. Has anyone else been able to get this all to work successfully? Looks pretty slick.

I have a couple of DID's through VP Connect and have been having sound quality issues on incoming calls. During the call, the calling parties voice sometimes sound like it is crackling, in other words it is not very crisp. I would liken it to listening to a radio with a blown speaker. This sound defect comes and goes throughout the call. The other person is always audible but it just isn't

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

... I just tried again after removing my hosts file entry (again) and outbound is now working! I had taken it out before, but I think I was getting a different error at the time. Sometimes it seems like asking for help is itself a cure! Thanks anyway! JDC

I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth

Asterisk v1.0 is running on RH 9. I installed festival RPM (festival-1.4.2-16.i386.rpm) and edited the festival.scm file to add: (define (tts_textasterisk string mode) "(tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is

I grabbed the latest sources from CVS yesteday and am having problems compiling. * v1.0.3 was running previously without issue. I tried checking out the older source but get the same make errors. The box is running RH 9. I am getting the following errors. Any thoughts on what is wrong? gcc -shared -Xlinker -x -o chan_mgcp.so chan_mgcp.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes

I have a couple of DID's with LiveVoip and am having major DTMF issues on incoming calls. I am connecting to them through IAX using ULAW. When someone dials one of these DD's (from a landline) they are for the most part unable to navigate the IVR menu successfuly. I would say the failure rate is greater than 80%. For example if the caller presses 5 sometimes * will see the DTMF as 55 or