search for: dingman

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 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

I have a model Q=K*A*(R^r)*(S^s) A, R, and S are data I have and K is a curve fitting parameter. I have linearized as log(Q)=log(K)+log(A)+r*log(R)+s*log(S) I have taken the log of the data that I have and this is the model formula without the K part lm(Q~offset(A)+R+S, data=x) What is the formula that I should use? Thanks for all of your help. I can provide a subset of data if necessary.

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

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

Is it possible to get an internal dial tone when I call to my asterisk and enter password? I would like to call my line enter extension - password - and get internal dial tone. once I'm in I would like to dial based on what context permits, mostly long distance calls VOIP. I can not preset the extension to certain number as I don't know what number I will be dialing. -- #Joseph

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

I'm trying to install spandsp But when I try to patch the Makefile it gives this error [root@pbxmilkshake apps]# patch < apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is it ok to go on

Hello, I want to add wireless capability to my Gentoo-linux based firewall/router at home, so I bought a Netgear MA311 PCI and installed the hostap package. I load the hostap_pci module and the wlan0 interface comes up fine. I can detect the signal from a wireless enabled laptop. Now I'm thinking I'm going to bridge the wlan0 interface and the eth1 interface, and run the firewall with br0

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

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

...sys.se> Subject: Re: [Asterisk-Users] fax e-mail spandsp To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <41E3BECD.4030003@upsys.se> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Matt Riddell wrote: > Brian Dingman wrote: > >> Anyone care to pass on a makefile that works. This is what my >> makefile.rej looks like: > > [SNIPPED] > > Really it's not that hard. Open two console windows. In one open > that patch. In the other open the Makefile. > > If you look at the p...

...info: [sip.conf] [general] port = 5060 bindaddr = 0.0.0.0 srvlookup=yes tos=lowdelay maxexpirey=3600 disallow=all allow=ulaw musicclass=default language=en relaxdtmf=yes ;useragent=Asterisk PBX ;nat=yes register => s00******:********@access1.voicepulse.com externip=asterisk.briandingman.com localnet=192.168.1.0/255.255.0.0 [voicepulse] type=friend context=voicepulse-incoming username=s00****** secret=******** host=access1.voicepulse.com dtmf=inband nat=yes qualify=yes canreinvite=no insecure=very [1000] type=friend host=dynamic ;callerid=Brian <1000> dtmfmode=rfc2833...