Re: [asterisk-users] AGI Delimiter in 1.6

2010-09-16 Thread James A. Shigley
. Then your second agi script which is basically the one that worked in 1.2/1.4 can use the channel variables. James Shigley -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Farmer Sent: Thursday, September

[asterisk-users] Asterisk Queue + Caller ID issue

2010-07-19 Thread James A. Shigley
there is caller ID? James Shigley -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello

[asterisk-users] Asterisk Queue + Caller ID issue

2010-07-19 Thread James A. Shigley
Let me rephrase this question. What context does a queue use for dialing out? James Shigley From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A. Shigley Sent: Monday, July 19, 2010 7:42 AM To: Asterisk Users Mailing

Re: [asterisk-users] Asterisk Queue + Caller ID issue

2010-07-19 Thread James A. Shigley
Could you give me an example because I understand what you said, but not sure what to put in my extensions.conf to accomplish that. James Shigley -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P

[asterisk-users] (no subject)

2010-07-16 Thread James A. Shigley
? James Shigley -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

[asterisk-users] Question

2010-02-24 Thread James A. Shigley
/111222LOCAL/222333,40) exten = 4095551212,n,Voicemail(1...@default) James Shigley -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Semi-Transfer

2010-02-03 Thread James A. Shigley
I've tried that as well prior to sending the initial email with no results. I'll play some with DISA today. James Shigley From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, February 02, 2010 2:50 PM

[asterisk-users] Semi-Transfer

2010-02-02 Thread James A. Shigley
out. Anyway so how would I accomplish this transfer of sorts? James Shigley Monroe Telephone Answering Service 409-981-9750 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments

Re: [asterisk-users] Semi-Transfer

2010-02-02 Thread James A. Shigley
That is the PRI span there are many available lines. James Shigley From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, February 02, 2010 2:12 PM To: 'Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Really Silly Question From Total Newbie

2010-01-05 Thread James A. Shigley
to forward to your asterisk server. from there you could use your dialplan to do whatever you wanted it to. And for outbound you would send the calls out thru the friends server via sip or iax. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104

[asterisk-users] SIP Issue

2009-12-28 Thread James A. Shigley
phone if no one answers in 21 seconds than it will roll over to that step. Any ideas? James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email

Re: [asterisk-users] SIP Issue

2009-12-28 Thread James A. Shigley
What do you mean I should use a global function. I'm kind both well versed and a newb to * James Shigley -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez Sent: Monday, December 28, 2009

Re: [asterisk-users] G729 Pass through

2009-12-11 Thread James A. Shigley
Have you paied for and imported g729 licenses from digium so that asterisks can use g729? http://store.digium.com/productview.php?category_id=5product_code=8G729 CODEC James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21

[asterisk-users] IAX2 Port issue

2009-12-04 Thread James A. Shigley
ideas? James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged

Re: [asterisk-users] IAX2 Port issue

2009-12-04 Thread James A. Shigley
192.168.16.3 is my desk 17.140 is * 192.168.16.0/21 is the subnet (255.255.248.0) Firewall isn't an issue here, that I can see for sure. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal

[asterisk-users] Variable Name needed

2009-12-02 Thread James A. Shigley
Supported: replaces, timer Contact: sip:+14092933193@ Content-Length: 0 Thank You for your time, and I apologize if this is a repeat question. I did Google, and search thru my * email archive (back thru April 09) for an answer first. James Shigley Monroe Telephone Answering

Re: [asterisk-users] Variable Name needed

2009-12-02 Thread James A. Shigley
That wasn't it either. I tried a few other likely fields from that page none of which gave the correct data James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE

[asterisk-users] FW: Variable Name needed

2009-12-02 Thread James A. Shigley
It might be worth mentioning the voip call is coming from a number we have thru bandwidth.com in case anyone uses them. James Shigley From: James A. Shigley Sent: Wednesday, December 02, 2009 3:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk

Re: [asterisk-users] Variable Name needed

2009-12-02 Thread James A. Shigley
Thank you that was it James Shigley Monroe Telephone Answering Service From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Wednesday, December 02, 2009 4:03 PM To: 'Asterisk Users Mailing List - Non-Commercial

[asterisk-users] Queue Question

2009-08-21 Thread James A. Shigley
to an interface/context. We are an answering service. If no agent is available in the queue I want to send it to the Interface which goes to my TAS equipment. So DAHDI/g2/Exten How do I accomplish that because I can't figure it out from googling or http://www.voip-info.org/ James Shigley

Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

2009-06-18 Thread James A. Shigley
the call, ourstate Null, peerstate Null !! Got reject for frame 71, but we have nothing -- resetting! James Shigley -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres Sent: Wednesday, June 17, 2009 3:20 PM

Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

2009-06-18 Thread James A. Shigley
I didn't have a limit set, but I put one on of 5 for testing sake that didn't change a thing. James Shigley Monroe Telephone Answering Service From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, June

Re: [asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

2009-06-18 Thread James A. Shigley
It errors the same whether I use g or G. James Shigley From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Thursday, June 18, 2009 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re

[asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

2009-06-17 Thread James A. Shigley
is 'CHANUNAVAIL' == Using SIP RTP CoS mark 5 -- Executing [9819...@from_test:1] Set(SIP/test-09f23d18, CALLERID(name)=James Shigley) in new stack -- Executing [9819...@from_test:2] Set(SIP/test-09f23d18, CALLERID(number)=4099819213) in new stack -- Executing [9819...@from_test:3

[asterisk-users] Asterisks, Sip to Local PRI/PTSN issue

2009-06-17 Thread James A. Shigley
) -- Auto fallthrough, channel 'SIP/test-b6369010' status is 'CHANUNAVAIL' == Using SIP RTP CoS mark 5 -- Executing [9819...@from_test:1] Set(SIP/test-09f23d18, CALLERID(name)=James Shigley) in new stack -- Executing [9819...@from_test:2] Set(SIP/test-09f23d18, CALLERID(number

Re: [asterisk-users] Bandwidth management and ADSL router

2009-05-26 Thread James A. Shigley
never used it in a virtual environment, but I see no reason why it wouldn't work that way. Also note that it requires almost nothing to run so you can put it on an old 1Ghz machine and It would still operate just fine. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC

Re: [asterisk-users] Ghost ??

2009-05-19 Thread James A. Shigley
can be causes by some many things the best place to start is to find a pattern if there is one. James Shigley Monroe Telephone Answering Service From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC Sent: Tuesday, May 19, 2009 1

Re: [asterisk-users] Ghost ??

2009-05-19 Thread James A. Shigley
Then most likely adam is right. You have interference/crossover on an analog line in your building or on the telco end. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps

Re: [asterisk-users] Is anyone keeping up with the versions?

2009-05-12 Thread James A. Shigley
Unless there is a new feature or your making a new system. Don't fix it if it aint broke. BUT do stay current on reading about new feature and things in the releases. James Shigley From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf

[asterisk-users] AGI PHP

2009-05-04 Thread James A. Shigley
works. No point making this any longer than need be. ? James Shigley ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

[asterisk-users] test

2009-04-30 Thread James A. Shigley
Had an inbound email server issue, just double checking it is working again. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any

Re: [asterisk-users] Feature request: manager show events

2009-04-24 Thread James A. Shigley
Then a suggestion for the next version would be to have a module which has the core set of events that are common to most everything for listing and added too, but still leave it open for the custom events most everyone uses for one thing or another. James Shigley Monroe Telephone Answering

[asterisk-users] AGI PHP script

2009-04-23 Thread James A. Shigley
I'm missing? Besides something not being configured in php.ini correctly any other ideas? James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email

Re: [asterisk-users] AGI PHP script

2009-04-23 Thread James A. Shigley
Actually I feel like an idiot. I had forgotten to put asterisk as an allowed sender in the server that those emails are going out of. (different from what * normally uses to email us) James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104

Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls usingAsterisk

2009-04-13 Thread James A. Shigley
What do you see when you run asterisk –r and dial 210 or 211 from one of the phones James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including

Re: [asterisk-users] Asterisk-beginner : cannot make phonecalls using Asterisk

2009-04-13 Thread James A. Shigley
Alright again, what do you see on the CLI when you make a call to 210/211? James Shigley From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Monday, April 13, 2009 12:07 PM To: Asterisk Users Mailing List - Non

[asterisk-users] IVR Survey

2009-04-10 Thread James A. Shigley
. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged

Re: [asterisk-users] IVR Survey

2009-04-10 Thread James A. Shigley
(s) to share.. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps,   CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged