similar to: Dialout to Meetme Fails?

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

I used the "FreePBX on Debian" HowTo at http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles to initiate calls to my SIP carrier. They get my registration, but they see that my call is interrupted before they can complete the connection. My Asterisk log shows that the call times out after the time (45s) specified in my dialplan Dial() command. What is wrong? [from

(Sorry if this has hit the list before. I sent it yesterday, but never saw it come through) I actually got rid of my phone service.. no more pots line in my house. But, I miss my call screen macro. Any way to do this with a SIP channel? (Obviously the parking isn't the problem, but rather recording their name). I set it up so they should only have to record their name once provided they

quesion 1: If I write extension.conf exten => 100,1,dial(sip/100,20,L(0)) I will listen ring but I don't access when other one dial 100. who have this experience ? quesion 2: and I look at this extension.conf in voip-info.org exten => _908.,1,Dial(Modem/ttyI0:${EXTEN:1}) Could this modem dial to PSTN? and will I install dial program for the modem? and when I write : exten

I actually got rid of my phone service.. no more pots line in my house. But, I miss my call screen macro. Any way to do this with a SIP channel? (Obviously the parking isn't the problem, but rather recording their name). I set it up so they should only have to record their name once provided they call from the same number. Anyway, I think I'm gonna get a big "NO" on

Does the SugarCRM included with AAH 1.3 not meet this criteria for you? ------------------------------ Message: 3 Date: Tue, 26 Jul 2005 17:48:08 +0100 From: "JunkMail" <junkmail@segurajuda.dyndns.org> Subject: [Asterisk-Users] Dial using URI(web) or using FORM(web) To: <asterisk-users@lists.digium.com> Message-ID: <009101c59201$cd5cc9c0$0a00a8c0@segurajuda.local>

Olivier, I think you are getting confused. Call me on 212-203-4357 and I'll answer your questions but basically I think you are doing this the wrong way. Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ________________________________ From: asterisk-users-bounces at

Hello! I have an Asterisk@home instalation with 7 users working OK, and I'ld like to implement either a -- Web dial feature, where the user would fill one form field with a phone number and a connection would be created between his extention and the entered number. OR -- Dial using an URI (callto:xxxxx link in a web page), having AstTapi installed and configured in all workstations.

Hi, I'm trying to get my dialplan to continue executing in the current context after a third-party is called and hangs up. It seems like it should be straightforward but it's not working. Here's what I have in extensions.conf: exten => 333,1,Answer() exten => 333,n,Playback(hello) exten => 333,n,Dial(SIP/19992223333 at sipcarrier,,g) exten => 333,n,Playback(hello) exten

Hi, Where can I find relevant information concerning callto:// tags ? Is it standardized or browser specific ? How within your browser, can you specify the software and parameters to used when clicking on such callto:// tags ? I couldn't find much googling or reading Preferences tab in Firefox. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Has someone successfully configured support of either callto or tel protocol in MS Office in general or MS Office Online's Outlook specifically ? (I'm referring here in Outlook client embedded in MS Office cloud service). If positive, what are the basic steps to enable such feature (clicking on a contact phone number triggers whatever program is attached to tel/callto protocol in

merge sirve pero no para cumplir la condición de si un dato es "x", buscarlo en el otro data.frame y asignarlo El mié., 20 mar. 2019 a las 10:23, Carlos J. Gil Bellosta (< cgb en datanalytics.com>) escribió: > ?merge > > El mié., 20 mar. 2019 a las 14:22, MAURICIO MARDONES (< > mauricio.mardones en ifop.cl>) escribió: > >> Amigos erreros >>

Toda la razon!!! merge era todo! Saludos El mié., 20 mar. 2019 a las 10:57, Carlos Ortega (<cof en qualityexcellence.es>) escribió: > Hola Mauricio, > > No, creo que no es lo que dices.. > > Con merge, indicas por qué columna (pueden ser varias) quieres juntar los > dos dataframes y con los parámetros "all.x", "all.y" y "all" indicas si

Hello all, Encouraged by the successful "demo", I'am getting on with Asterisk CVS. I added 2 H.323 extensions in extensions.conf [default] include => demo exten => 701,1,Dial(H323/gm2@192.168.1.20/s) exten => 702,1,Dial(H323/gm2@192.168.1.25/s) With: - [demo] is defined by default in sample.extensions.conf - Asterisk server is running on host 192.168.1.20, on the

Hi all, I try to ask helps again about chan_h323 extensions. I define this in h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 allow=gsm allow=ulaw gatekeeper = DISABLE context=default [gm1] type=friend host=192.168.1.20 context=default [gm2] type=friend host=192.168.1.25 context=default and I have in extensions.conf : [demo]

Heelo, I currently search a program that can make a web browser Pop-up on an incoming call on a specific URL like : http://directorie.ch?CALLNUMBER:00451849799 I have found ADM, but it's a bit more complex for my purpose an it's not very stable. Do you know a simple software for that ? An other part of my project is to eneable click-to-call from a web page, do you know a kind of

*Q: Quick question: does "battery.runtime" update when the device is on battery, or is this field always zero?* *A: Yes, the UPS battery time starts counting when the device starts on battery.* Q: "battery.runtime.low" should be the number of seconds remaining before switching to "LB" status. This seems large, even for a 6000 VA UPS. Is this configurable? A: I

Aquí hay una posible solución... https://tex.stackexchange.com/questions/27138/how-can-i-fix-the-error-gui-framework-cannot-be-initialized-with-texniccenter El mié., 19 feb. 2020 a las 23:16, MAURICIO MARDONES (< mauricio.mardones en ifop.cl>) escribió: > Esto... > > 2020-02-19 18:29:50,307-0300 INFO pdflatex - starting with command line: >

Hi, we have implemented a first version of call support from a web based system for Asterisk (via the manager interface) and other, callto: and tel:, providers. Now I am looking at the other way around. If a call comes in, I want our web based system to automatically detect the number and present the call information to the user. Ideas anyone? I guess, I won't be able to get this done

hi, I started today to use the acts as taggable plugin and it''s working as expected. I added a method find_tagged_with_all since I needed to find the elements tagged with all the items in a list while find_tagged_with finds all the elements tagged with any of the list terms. Attached to this mail there''s a patch which adds the find_tagged_with_all method and which solves a