[asterisk-users] why not forwarding to this number?

2014-08-09 Thread Thomas Perron
exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,SayAlpha(495256) exten = s,n,Wait(2) exten = s,n,Dial(SIP/222) exten = s,n,Hangup -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] why not forwarding to this number?

2014-08-09 Thread Thomas Perron
hi resolved. added: include=outgoing cheers On Sat, Aug 9, 2014 at 7:34 AM, Administrator TOOTAI ad...@tootai.net wrote: Le 09/08/2014 12:23, Thomas Perron a écrit : exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,SayAlpha(495256) exten = s,n,Wait(2) exten = s,n,Dial(SIP

[asterisk-users] (no subject)

2013-04-12 Thread Thomas Perron
Basic Dial Plan Why is this plan not engaging the line exten = 105,n,Dial(SIP/voipvoip.com/1703501) and dialing the 703 number? The logs and debug dont show any problems [incoming] exten = 44,1,Answer() exten = 44,n,Wait(1) exten = 44,n,Playback(beep) exten =

[asterisk-users] Connect to an outbound channel and dial a phone number??

2013-04-09 Thread Thomas Perron
This seems basic but something is missing. I dial from my cell phone to my DID and enter the context in extensions.conf I am hoping to cascade through the plan and successfully automatically dial the 1444 number listed. But it fails. And, I dpon't know why? Should I removed the Hangup

[asterisk-users] extensions.conf / test DID

2013-04-08 Thread Thomas Perron
I am trying to make sure my DID and SIP account details are working properly and engaging the extensions.conf and dial plan. I have a successful SIP session registered: Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922) Asterisk*CLI sip show registry Host

Re: [asterisk-users] sip registration

2013-04-07 Thread Thomas Perron
Edwards asterisk@sedwards.comwrote: A better subject will yield better replies. On Sat, 6 Apr 2013, Thomas Perron wrote: Shouldnt I be able to at least ping the SIP provider IP? Not if they don't allow it. They don't. sip3.voipvoip.com registers fine for me with your credentials

[asterisk-users] sip registration

2013-04-06 Thread Thomas Perron
I have a very lite layout and attempting to get the SIP configuration set up initially before proceeding into other areas. VMware is running my Asterisk 11 on Ubuntu 12. Shouldnt I be able to at least ping the SIP provider IP? I run command sip show registry and do not see it set up. I run sip

[asterisk-users] Rookie / sip and extensions

2012-07-07 Thread Thomas Perron
Sorry for blasting another desperate note but I am trying! I have changed the username and password and IP to protect my system. But, the logic is unchanged. It is does not work! I simply want to dial the telephone number provided to me for my DID which corresponds with my SIP info. And, then

[asterisk-users] sip and extensions

2012-07-05 Thread Thomas Perron
I am new. Here is the code that I am playing with on CentOS 6.x When I dial the number that corresponds w/ my SIP account I get a recording: reached a non-working number I built Asterisk a few times last year and am now back working on a similar project. In my view, there is

Re: [asterisk-users] sip and extensions

2012-07-05 Thread Thomas Perron
Hi, I changed these codes to not coincide with actual account info. Thanks On Thu, Jul 5, 2012 at 5:48 PM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - I am new. Here is the code that I am playing with on CentOS 6.x register =

[asterisk-users] basic sip quesiton

2012-07-04 Thread Thomas Perron
What am I missing please? sip show registry shows that I am registered. [general] register = 5552530146:tam...@sip3.voipvoip.com ; ; [sip3.voipvoip.com] bindport=5060 ;you can use different port if the default is blocked bindaddr=0.0.0.0 ;binds to all ;this is for codec negotiation between

[asterisk-users] call files .vbs

2011-05-22 Thread Thomas Perron
This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses but I want to know in any case! Can a vb script run somehow on a Linux machine or does it only work on Windows? If I were to build a call file script (described in this link

Re: [asterisk-users] call files .vbs

2011-05-22 Thread Thomas Perron
Hi Doug, Yes. I have sorted that part out. Also, it seems like the pscp function is the way that I can tie together the vb script with the logic of the Asterisk call files learning curve!! Thanks On Sun, May 22, 2011 at 8:37 PM, Doug Lytle supp...@drdos.info wrote: Thomas Perron wrote

[asterisk-users] DHCP / DNS

2011-04-27 Thread Thomas Perron
Are there any internal DHCP or DNS services built-in to the Asterisk code? -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

[asterisk-users] IP Address Management / Open Source / IPAM

2011-04-19 Thread Thomas Perron
Does anyone have a recommendation for an Open Source IP Address Management solution please? There are several commercial players such as BlueCat, BT Diamond, InfoBlox, VitalQIP. And, Solarwinds makes a module that focuses on IPAM. Most vendors tie logic into DNS and DHCP into IPAM designs. In

[asterisk-users] console debugging

2011-01-28 Thread Thomas Perron
I used the command asterisk -vc to see console messages and it works fine. Now, I want to turn off this feature. How please? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] console debugging

2011-01-28 Thread Thomas Perron
OK. That is what I needed to know. Thannks On Sat, Jan 29, 2011 at 12:34 AM, Warren Selby wcse...@selbytech.com wrote: On Fri, Jan 28, 2011 at 8:59 PM, Thomas Perron thomas.per...@gmail.com wrote: I used the command asterisk -vc to see console messages and it works fine. Now, I want

Re: [asterisk-users] Basic Sip.conf and extensions.conf

2011-01-17 Thread Thomas Perron
Thanks. I fixed that. Still does not work. On Mon, Jan 17, 2011 at 12:53 AM, Jeroen Eeuwes jeroeneeu...@gmail.com wrote: Hi Thomas, register = 999:999...@sip.callwithus.comi Perhaps this should be .com instead of .comi ? Best regards, Jeroen Eeuwes --

[asterisk-users] Basic Sip.conf and extensions.conf

2011-01-16 Thread Thomas Perron
Does anyone see any issues here? I cannot get it to work. Passwords are not real! [general] ;register = 999:999...@carrier.callwithus.com register = 999:999...@sip.callwithus.comi context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes ; enable DNS SRV server [joesipshow] type=friend

Re: [asterisk-users] Basic Sip.conf and extensions.conf

2011-01-16 Thread Thomas Perron
OK. I set up the logger.conf via the steps provided. Now, how do I get the results. I reproduced the scenario. On Sun, Jan 16, 2011 at 4:02 PM, Paul Belanger pabelan...@digium.com wrote: On 11-01-16 03:58 PM, Thomas Perron wrote: Does anyone see any issues here?   I cannot get it to work

[asterisk-users] incoming

2011-01-02 Thread Thomas Perron
Is it possible to have Calls incoming to different DIDs? I want an AA that handles 100s of businesses. [Incoming-pizza] Exten = 4045551212,1,Goto(pizza,s,1) [Incoming-hvac] Exten = 8085551212,1,Goto(hvac,s,1) [Incoming-gutter] Exten = 6175551212,1,Goto(gutter,s,1) --

Re: [asterisk-users] incoming

2011-01-02 Thread Thomas Perron
BOX 811061, Chicago, IL, USA, 60681. www.readywire.com. On Sun, Jan 2, 2011 at 11:50 AM, Thomas Perron thomas.per...@gmail.comwrote: Is it possible to have Calls incoming to different DIDs? I want an AA that handles 100s of businesses. [Incoming-pizza] Exten = 4045551212,1,Goto(pizza,s,1

[asterisk-users] Mail Integration

2010-12-13 Thread Thomas Perron
Does anyone have a super simple cookbook describing the steps to integrate Mail into an Asterisk Dial Plan. I have googled but have a lot of choppy results. I am running RH and Asterisk 1.8 Cheers Tom -- _ -- Bandwidth and

Re: [asterisk-users] Mail Integration

2010-12-13 Thread Thomas Perron
...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: Monday, December 13, 2010 5:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Mail Integration Does anyone have a super simple cookbook describing the steps to integrate Mail

[asterisk-users] debug audio or channel

2010-12-07 Thread Thomas Perron
Does anyone have any short answers on how I can fix this problem: A calls B. B rings Says connected. But the call is not bridged and therefor no audio passes. very simple dial plan. Frustrated. v 1.8 -- _ -- Bandwidth and

Re: [asterisk-users] Dahdi issue with Asterisk 1.8.0

2010-12-07 Thread Thomas Perron
Do you have any issues with getting audio to bridge? I am using 1.8 also. On Tue, Dec 7, 2010 at 12:38 PM, Timothy Legge timle...@gmail.com wrote: Hi I was using the delivered Ubuntu 1.6.x packages but I wanted to look at gtalk integration so I downloaded, compiled and installed the source

[asterisk-users] no audio on end-point when call is connected/bridged via PBX

2010-12-06 Thread Thomas Perron
I am trying to dial through my asterisk machine from phone A to phone B. My DID is registered properly with the SIP provider. When I dial from A to B it looks fine so far. A rings B and B can pick up and the call is bridged. However, I don't hear any audio so therefor it is not working. I am

[asterisk-users] no audio

2010-12-05 Thread Thomas Perron
Any reason why I don't get audio on the channel after it rings and the end user picks up. Here are my files. CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] include = stdexten exten = s,1,Answer() exten = s,n,Wait(1) exten =

Re: [asterisk-users] no audio

2010-12-05 Thread Thomas Perron
(1) exten = s,n,Dial(SIP/callwithus/44) exten = s,n,Wait(2) exten = s,n,Hangup() ~ On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 5 Dec 2010, Thomas Perron wrote: Any reason why I don't get audio on the channel after it rings and the end user

Re: [asterisk-users] dial plan and sip

2010-11-15 Thread Thomas Perron
thank you i will try it. On Mon, Nov 15, 2010 at 4:52 PM, Chad Wallace cwall...@lodgingcompany.com wrote: On Sat, 13 Nov 2010 20:38:30 -0500 Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config.  But, not working (: I simply want to dial the DID that is registered

[asterisk-users] dial plan and sip

2010-11-13 Thread Thomas Perron
Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register =

Re: [asterisk-users] dial plan and sip

2010-11-13 Thread Thomas Perron
...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config.  But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial

[asterisk-users] upgrade

2010-11-13 Thread Thomas Perron
i am running 1.4.37 and am hosted on Rackspace. I feel like a took a step back by using the Cloud server service since I am having a little trouble proving that my basic configuration is working. Nevertheless, I want to upgrade to 1.8. I use Centos 5.5 Anyone know of a good link that can help

Re: [asterisk-users] dial plan and sip

2010-11-13 Thread Thomas Perron
Woollum br...@woollum.com wrote: What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett

Re: [asterisk-users] dial plan and sip

2010-11-13 Thread Thomas Perron
execute. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 13, 2010, at 7:02 PM, Thomas Perron wrote: How do I see the error message? the phone call seemed to get through but I did not see anything on my 1.4 console. i used 1.6.x before.  having trouble

[asterisk-users] install

2010-11-07 Thread Thomas Perron
I have installed Asterisk before w/ no issues but while trying today (1.6.2.13 and centors 5.4) I receive the following at the CLI: The configure script must be executed before running 'make'. Please run ./configure. Any tricks on getting through this? I did not select to

Re: [asterisk-users] Registering and initiating a SIP call without a SIP client

2010-09-05 Thread Thomas Perron
Yes. Send your code. Consider using call files. Here is a part of what works for me. [-system] exten = s,1,Answer exten = s,n,Wait(2) exten = s,n,Playback(pa-welcome) please record your broadcast after the beep ;exten = s,n,Playback(beep) exten = s,n,Wait(1) exten =

[asterisk-users] fast busy out?

2010-09-04 Thread Thomas Perron
why does this not work? i simply want to hear the recorded message exten = s,1,Answer() ;exten = s,n,Record(zipcodegutter1.gsm) ;zcg1 exten = s,n,Playback(zipcodegutter1) exten = s,n,Dial(SIP/c01s/159,120,A,(demo-thanks)) --

Re: [asterisk-users] fast busy out?

2010-09-04 Thread Thomas Perron
...@gmail.com: no I am not sorry, and please reply to this list, and not to me directly.. On Sat, Sep 4, 2010 at 6:16 PM, Thomas Perron thomas.per...@gmail.com wrote: thank you for your note on the Asterisk users group list Are you in Scandanavia somewhere? Cheers Tom -- -- Ondrej Škopek

[asterisk-users] append CID label

2010-06-26 Thread Thomas Perron
I want a call to connect via my DID to my dialplan. Then, I want to attach a label to the incoming call call arrives starts to dive through the dial plan then rings a trunk/channel via SIP (see below) Question: before answering my 1212111 endpoint I want to see a flag CID that correlates to

Re: [asterisk-users] append CID label

2010-06-26 Thread Thomas Perron
ok thank you i will try On Sat, Jun 26, 2010 at 10:31 PM, C F shma...@gmail.com wrote: exten = s,n,Set(CALLERID(name)=label${CALLERID(name)}) put this before the dial command. On Sat, Jun 26, 2010 at 10:09 PM, Thomas Perron thomas.per...@gmail.com wrote: I want a call to connect via my

Re: [asterisk-users] MeetMe problem

2010-06-12 Thread Thomas Perron
try using confbridge in lastest asterisk version On Sat, Jun 12, 2010 at 11:30 AM, Daniel Knoll dan...@danielknoll.de wrote: Hi Guys, sometimes if one caller or many callers are in a meetme Room and a new one join the room, then he or another caller into the same room where kickt from the

[asterisk-users] text

2010-05-07 Thread Thomas Perron
Does anyone know how to send a text message from Asterisk? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] text

2010-05-07 Thread Thomas Perron
...@txt.att.net) On Fri, May 7, 2010 at 8:32 PM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 7 May 2010, Thomas Perron wrote: Does anyone know how to send a text message from Asterisk? Carrier specific, but how about:        system(echo foo | mail -s bar 551...@txt.att.net

[asterisk-users] meetme / upgrade to 1.6.2.6

2010-04-18 Thread Thomas Perron
I read that I need to run 1.6.2.6 (at least 1.6) to use the MeetMe application since I don't have a zdummy timing driver. Anyway, I want to upgrade my machine to 1.6.2.6. Does anyone have the exact steps? I see a lot of references on the web but any other links from our community may be preferred!

[asterisk-users] migration

2010-03-27 Thread Thomas Perron
My client wants to use my service that I will host. It is an IVR application. I have the solution worked out on the server side. I will port his current POTS line phone number to a DID service where I can control it via SIP. Question relates to his current phones. Forgive me as I am new. Does

Re: [asterisk-users] Play music to caller after answer, before dial

2010-03-22 Thread Thomas Perron
Does this help? The A near the end calls the audio file ginr3 exten = 551,1,Answer() exten = 551,n,Dial(SIP/callwithus/17025551212,120,A(ginr3)) On Mon, Mar 22, 2010 at 6:41 PM, Michelle Dupuis mdup...@ocg.ca wrote: I think I forgot some important information... I'm actually running an AGI

[asterisk-users] queue MOH

2010-03-14 Thread Thomas Perron
I want callers to enter a queue and then hear music on hold. does anyone have notes on how to integrate queuing to a dial plan that uses moh? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

[asterisk-users] DID forwarding ?

2010-03-13 Thread Thomas Perron
DID number A. I have a DID (a regular line from Verizon). number A. Can I have A ported to my SIP provider? Then, interface the A DID to my system so that I can build a solution. I want to write an IVR for the A number and allow callers dialing A to interact with my Asterisk machine. I need to

[asterisk-users] IVR Demo / Create file / Move file / Demo all

2010-02-08 Thread Thomas Perron
Do you see any syntax errors? Positive comments welcomed. The short version of the logic is as follows: create a file based on the NUMBER create a an audio file move to a new extension (label) and play the results exten = 621,1,Answer() exten =

Re: [asterisk-users] IVR Demo / Create file / Move file / Demo all

2010-02-08 Thread Thomas Perron
Demo / Create file / Move file / Demo all On Mon, 8 Feb 2010, Thomas Perron wrote: Do you see any syntax errors? Yes. Lots. Can I please have the last 5 minutes of my life back? Positive comments welcomed. Please don't bother the list to syntax check your code if you are too lazy to type

Re: [asterisk-users] IVR Demo / Create file / Move file / Demo all

2010-02-08 Thread Thomas Perron
= 621,n,Playback(snowday2) exten = 621,n,Goto(s,1) On Mon, Feb 8, 2010 at 2:00 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Feb 08, 2010 at 12:36:18PM -0500, Thomas Perron wrote: what is OP please? can you just simply comment on the technical work please? Original Poster. The one

[asterisk-users] syntax

2010-02-07 Thread Thomas Perron
I am trying to understand .call files. The logs seems to indicate issues with the audio file that I am trying to have played when the call is connected. Any thoughts? Some sample files and logs to console are shown. zipp-code.call Channel: SIP/callwithus/12023519259 Application: Playback

Re: [asterisk-users] syntax

2010-02-07 Thread Thomas Perron
thing is that you may not have to put the file extension in the name if the file is in the proper place as well. Try that and see what happens. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas

Re: [asterisk-users] Dial script

2010-02-06 Thread Thomas Perron
My inquiry is to understand how I could configure a system to do it. I have since learned that Asterisk has features in the code to do this (auto dial out, features.conf and .call files.) The 1 example is a bit extreme but it really does not matter what the number is for this. Dialogic has

Re: [asterisk-users] Dial script

2010-02-06 Thread Thomas Perron
- From: Thomas Perron thomas.per...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 06, 2010 4:56 AM Subject: Re: [asterisk-users] Dial script My inquiry is to understand how I could configure a system to do

Re: [asterisk-users] Dial script

2010-02-06 Thread Thomas Perron
of this discussion harms the Asterisk community as a whole. with friendly regards, Erik de Wild Tripple-o: your asterisk migration partner the Netherlands On 6 feb 2010, at 03:54, Thomas Perron wrote: Does anyone have a Dial script or a hint on how I can dial 1 numbers

[asterisk-users] Dial script

2010-02-05 Thread Thomas Perron
Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- Bandwidth and

Re: [asterisk-users] Dial script

2010-02-05 Thread Thomas Perron
karl, does it make you feel good ? wow. pathetic. On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife karlf...@gmail.com wrote: Try this: #rm -rf / - Original Message - From: Thomas Perron thomas.per...@gmail.com To: asterisk-users@lists.digium.com Sent: Friday, February 05, 2010 8:54 PM

[asterisk-users] MATH

2010-02-02 Thread Thomas Perron
I want to allow users to dial my DID Then, hear my ginger3 intro Then, depending on the number that they press, provide a total via MATH. Comments. Will this work? exten = 866,1,Goto(tommath,s,1) [tommath] exten = s,1,Read(NUMBER,ginger3,2,skip,5) exten = s,n,Gotoif($[${NUMBER} = 14]?onefour)

Re: [asterisk-users] MATH

2010-02-02 Thread Thomas Perron
hi Steve, I am trying it and I am using the feedback from the group. In my view, that is the purpose; try, test, talk. Thanks for your interest. On Tue, Feb 2, 2010 at 7:15 PM, Steve Edwards asterisk@sedwards.com wrote: On Tue, 2 Feb 2010, Thomas Perron wrote: I want to allow users

Re: [asterisk-users] MATH

2010-02-01 Thread Thomas Perron
-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Sunday, January 31, 2010 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MATH On 31/01/10 6:27 PM, Thomas Perron wrote: what is wrong

Re: [asterisk-users] MATH

2010-01-31 Thread Thomas Perron
possible commands) 2010/1/31 Håkon Nessjøen haa...@avelia.no: You probably have to do a exten = s,1,n,Set(TOTAL=0) in the start of the call, to initialize the TOTAL variable On Sun, Jan 31, 2010 at 4:29 AM, Thomas Perron thomas.per...@gmail.com wrote: thanks for the response. I tried

Re: [asterisk-users] MATH

2010-01-31 Thread Thomas Perron
ok. that worked thanks!! On Sun, Jan 31, 2010 at 10:50 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Jan 31, 2010 at 10:37:29AM -0500, Thomas Perron wrote: hi i don't claim to be a star at this but there must be some obvious part missing; my dial plan is below.  out put from

Re: [asterisk-users] MATH

2010-01-31 Thread Thomas Perron
does dtmf any any variable that i can capture and use w/ some logic like in the case of a gotoif so, if caller enters a certain number then gotoif matches XX otherwise go to YY. On Sun, Jan 31, 2010 at 10:58 AM, Thomas Perron thomas.per...@gmail.com wrote: ok. that worked thanks!! On Sun

[asterisk-users] MATH

2010-01-30 Thread Thomas Perron
I want to create a script for IVR that compiles responses, aggregates them to a total number. Then, run an equation based on the result. Press 1 for X (X is a positive number 500) Press 2 for Y (Y is a positive number 200) Press 3 for Z (Z is a positive number 300) Press 20 to calculate the

Re: [asterisk-users] MATH

2010-01-30 Thread Thomas Perron
total up for current call. then read back the number 2010/1/30 Håkon Nessjøen haa...@avelia.no: For all calls combined, or for the current call? On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron thomas.per...@gmail.com wrote: I want to create a script for IVR that compiles responses

Re: [asterisk-users] MATH

2010-01-30 Thread Thomas Perron
used either math or saynumber before, but according to the documentation, something like this should work.. On Sat, Jan 30, 2010 at 3:06 PM, Thomas Perron thomas.per...@gmail.com wrote: total up for current call. then read back the number 2010/1/30 Håkon Nessjøen haa...@avelia.no

[asterisk-users] MATH