search for: crompton

...ther large. Maybe 50-100. The idea is that I can talor my local PSTN usage to only those exchanges that are non-toll for me. Other calls will go out ENUM to best way. So is there a way to do this without entering a dialplan for every exchange designator? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

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

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 * ****************************

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 * ****************************

...this? I am not using any dial flags. I tried 'inband' in all places with no difference. At one point this seemed like a * feature problem and I thought removing dial flags fixed it but that does not now seem to be the case. How can I fix this??? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

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" <asterisk-users@lists.digium.com> Sent: Monday, February 12, 2007 10:33 PM Subject: [asterisk-users] Using Asterisk/callerid with "pay as you go" VOIPproviders >I am curiou...

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

...quot; ; echo " attempting to run Asterisk." ; echo "" ; for f in ; do echo " $f" ; done ; echo "" ; echo " WARNING WARNING WARNING" ; fi' make[1]: *** [oldmodcheck] Error 2 make: *** [install] Error 2 **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

I do a ./configure successfully but when I try doing a 'make' I get error 1 - menuselect What am I doing wrong? Doug

...include gvrp include stp::rstp stp::portfast { "1/e1-e48,2/e1-48",3/e1-48": } } Now I know we probably can''t get puppet to run on the switch, but we can get a host to ssh or telnet to the switch, and to download the current configuration of the switch. -- +-Geoff Crompton +--Debian System Administrator +---Trinity College -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this group, send email to puppet-users@googlegroups.com. To unsubscribe from this group, send email to puppet-users+unsubscrib...

...ke to be able to use features but the ability to have reliable DTMF after call completion is more important. This is not a problem with the default transfer key "#" as it happens no matter what it is defined as. Anyone have any fixes for this? Doug **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

...val 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 * * * * doug@crompton.com * * http://www.crompton.com * ****************************

...ht way to use filebucket to do this, or if there is something else I should be reading up on. My end goal for this is to build a /var/lib/misc/passwd.db and group.db on the puppet master (using nss_updatedb, from ldap), and then push out those .db files to the puppet clients. Thanks! -- +-Geoff Crompton +--Debian System Administrator +---Trinity College --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this group, send email to puppet-users@googlegroups.com To unsubscribe from...

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

...t; To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > "Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety." -- Ben Franklin (1759) **************************** * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307 * * * * doug@crompton.com * * http://www.crompton.com * **************************** _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIB...

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 =>