similar to: Compile install error.

Being rather new to Asterisk I was wondering what the current status of AEL2 is? I see reference to it back in January but that was many versions ago. Is it in the current code? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

Ok asked this earlier with no response so I will phrase it a different way. I am sure someone had to deal with this and there is a "best way." I want to let Asterisk make the decision on best path based on local exchange - xxx-yyy - where xxx is one of my local area codes and xxx is exchange designator. The problem is that the list is rather large. Maybe 50-100. The idea is that I can

Does it matter if you use upper or lowercase rules - I.E. - "x" vs. "X" or mix them? Not that I would do that as a rule but sometimes you make mistakes! Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

Ok trying this again... is there anyone using the SPA-3000 with * ???? I am not sure if this is a specific problem to it or not. This is something I really need to fix!!! When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot access (reliably) DTMF menus at the called party, after call completion. Dialing DTMF is fine. I checked by calling myself. Listening to either end on a

Recently started using * and really am having fun. One problem I encountered... I am using an SPA-3000 3.1.10d When I have transfer enabled - 'T' in the dial string I cannot reliably send DTMF keys to a bank, voicemail, or other service requiring tones. If I disable (remove transfer option) from the dial string all is fine. I would like to be able to use features but the ability to have

Not being very happy with festival I would like ro get a better TTS engine. I looked at the listings at: http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international but I would like to get user input on suggested packages for Linux. Best performance vs. cost ???? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * *

The problem was that autofallthrough=yes was set in extensions.conf > I'm experiencing a strange problem. My inbound calls are hanging up > right after Background() message even though response timeout is set to > 10 sec. > > [voicepulseincoming] > > exten=>_X.,1,Answer > exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1) >

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

If you asked this question on the biz list you would get a lot of people that will tell you that they offer services where you can set the caller ID to what ever you want. To name a few:: Nufone Teliax Voipjet ----- Original Message ----- From: "Doug Crompton" <doug@crompton.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Hi all, I need to simply use Asterisk to receive incoming calls in an IVR manner. It should authenticate users and read data from MySQL table that match their ID through Text-to-speech. I already have Asterisk 2.6 (Asterisk@home). I understand that I need to use Festival and AGI but do not know what to do exactly. Any help is appreciated. Thanks

Hi, I have a system that connects to the PSTN. What do I need to do so that when a call comes in, the system will start ringing the hunt group I have setup but not actually answer the call? The problem is the system is answering the call, and then passing 'ringing tones' back to the caller, so this makes the phone companies call-forward-no-answer not work since the telco thinks they

Three months ago, I was experiencing all sorts of issues with my Asterisk box maintaining a connection to multiple trunks, etc. I also experienced various timing issues as well. In addition, Asterisk would sometimes take almost a minute to fully load and register its SIP and IAX trunks. Puzzled, I recompiled several times. No result. I checked my hardware. Didn't find anything. However, I did

Hello, Before throwing in the towel with my Sipura 3000 has anyone had much success with that adapter connected to a door phone? In our setup a doorphone is connected to the SPA's fxs port. When a visitor rings, asterisk calls a group of Polycoms and the person who answers has to enter *1 to trigger the door opening. However it seems the SPA doesn't relay the DTMF's to the

Hi list, I tried to use DISA in order to get the line when I call with my mobile phone but the system doesn't recognise my DTMF tones when I call to a SIP trunk. Everything is working Ok if I use a ZAP Trunks. I tried to google to find a solution but I wasn't able to find any. Any idea? I'm using trixbox 2 with 4 SIP trunks and a Zaptel TDM 400 card. Bye

All - this is probably a simple problem, but I've been pulling my hair out trying to figure out what I'm doing wrong. I'm building a *simple* IVR menu. Here it is: [main-menu] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout(5) exten => s,4,ResponseTimeout(30) exten => s,5,Background(logic-main) exten =>

Is what I am trying to do in this context possible. That is changing the incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not preceeded by a "1" I want to add a "1". Often calls come in without the preceeding "1" and this plays havoc with my redial if the 3 digit area code matches a local 3 digit extension. All my outside calls are 10 digits or 1+10

I'm experiencing a strange problem. My inbound calls are hanging up right after Background() message even though response timeout is set to 10 sec. [voicepulseincoming] exten=>_X.,1,Answer exte=>_X.,n,GotoIfTime(9:00-17:00|mon-thu|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(9:00-15:00|fri|*|*?business-hours,s,1) exten=>_X.,n,GotoIfTime(*|*|*|*?after-business-hours,s,1)

My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: --------- In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

Ok, I am back from my thanksgiving holiday, and I find there was a big snow storm here in Seattle. Apparently during the storm there where multiple brown out/black outs. I have struggled since day one to get a high quality PSTN gateway configured with my very long loop and Mac based asterisk. I originally tried the HT-488, which had multiple issues, and was unacceptable. I then purchased