similar to: Send DTMF's before the call is answered

I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through Asterisk, and based on them, build the destination PSTN number). My problem is that Dial send the DTMF's to the SIP/gateway user at a rate

I'm developing an application that monitors the state of the incoming calls using Manager events. So, as a part of it, I need to "override" the control of the extensions by the dialplan itself. The problem is that, if I don't declare the incoming extension, Asterisk hangs up the call by default. So I want to know if there's some kind of "ManagerControl() application

I've set voicemail almost successfully, only a minor detail remains :-) I can't get the dates in my local language (spanish). In sip.conf, zapata.conf and voicemail.conf, I've set: language=es and my locale is "es" also. However, the days and months names still appear in english in the emails!!! Thursday 11 de May de 2006, 18:49:34. instead of Martes 11 de mayo de

I'm trying to set the Pickup feature. I'm setting my extensions.conf as: exten => _*.,1,Pickup(SIP/${EXTEN:1}) but if, for example, extension 03 is ringing by a call made from extension 01, and I try to pick it up from extension 02 (by dialing *03 from extension 02), I can see in the Asterisk console (Verbosity set to 10): -- Executing Dial("SIP/01-512c",

Is there a way to access only certain positions in the $EXTEN variable? I'd like to filter my international calls based on the destination country: My dialplan looks like this (1XX0. is the international calling convention for Chile) exten => _1XX0.,1,Dial(SIP/${EXTEN:4}@external_provider) But, I'd like to, depending on the destination country (digits 5 and eventually 6 of EXTEN),

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

I've defined my dialplan as showed below. My internal lines are numbered as 12345XX, and internal users can call another by the entire 7-digits extension, or by just last 2 digits. [invalid] exten => _X.,1,Playback(pbx-invalid) exten => _X.,2,Hangup() [internal] include => invalid exten => _XX,1,Dial(SIP/12345${EXTEN}) ; Short alias for internal lines exten =>

I wonder if someone could explain why, when I perform confirmatory factor-analysis model using polychoric correlations why I do not get an estimated confidence interval for the RMSEA. My experience with these type models is that I would obtain a confidence interval estimate. I did not get any warning messages with the output. RESULTS: Model Chisquare = 1374 Df = 185 Pr(>Chisq) = 0

Hi all, I'm trying to insert a Rhino Ceros box equipped with a Rhino R2T1 dual-T1 card and running the latest version of Trixbox (2.4.2) between the central office and a Nortel Option 11. The switch at the CO is a DMS100. Basically, I'm taking the T1, connecting it to port 0 on the R2T1 card, and then connecting port 1 to the Nortel. (Actually a CSU and then the Nortel) We're running

Hey Nithya, root at PB-WA-AA-00-A:/# glusterfs -V glusterfs 3.10.1 Repository revision: git://git.gluster.org/glusterfs.git Copyright (c) 2006-2016 Red Hat, Inc. <https://www.gluster.org/> GlusterFS comes with ABSOLUTELY NO WARRANTY. It is licensed to you under your choice of the GNU Lesser General Public License, version 3 or any later version (LGPLv3 or later), or the GNU General Public

Kind of spawns an interesting side topic though..... I recommend SpamHaus.org for a good blacklist.... Easy to integrate into most mail servers and you can't beat free... Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Matt Sent: Thursday, June 09, 2005 10:40 AM To: Asterisk Users Mailing

All the MIPS buildbots are ready too. From: llvm-dev [mailto:llvm-dev-bounces at lists.llvm.org] On Behalf Of NAKAMURA Takumi via llvm-dev Sent: 25 May 2016 23:03 To: Chris Bieneman; llvm-dev at lists.llvm.org; cfe-dev at lists.llvm.org; lldb-dev at lists.llvm.org Subject: Re: [llvm-dev] [Attn: Bot Owners!] Raising CMake minimum version to 3.4.3 I am ready, regarding to, http://bb.pgr.jp/ On

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1

you can not ping that address because ICMP is turned off. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Deon Rodden Sent: Monday, July 26, 2004 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Broadvoice problems again Attn: James Greetings, C:\>ping 147.135.8.129 Pinging

After many hours today thinking that I had placed a bug into my dialplan, I realized that for some reason DTMF tones are simply not making it into asterisk! Calling into my pbx transmits crystal-clear audio in both directions. But dialing DTMF's from pstn->pbx is unsuccessful, while pbx->pstn works fine. The tones simply don't make it through. Tiny brief fragments are all. Please

Hello everybody, I need help, I have a problem with conferences in asterisk, when many people are in a conference sometimes there're users pressing phone keys and this action emits a sound (DTMF of the phone keys), so, I need to find the way of not listening this sound.. I'm using MeetMe(variable,pFX).. I tried whithout "F" but it doesn't work because users continue

Hi, I've looked around the archives and have spent a while on voip-info.org but not found an answer so forgive me if this is in a FAQ somewhere. We've got several Asterisk servers with E1 cards (some Digium, some Sangoma). We provide non geographic numbers for customers and route calls to their existing phone numbers. Calls come in over the PSTN and into Asterisk. This works perfectly

I am using an example posted in this help forum to work with a file. the head of the file looks like: 988887 2007-03-05 2007-06-01 90 3 5.450 205500.00 999.00 999.000 0.000 0 0 988887 2007-03-06 2007-06-01 90 3 5.450 205500.00 999.00 999.000 0.000 1 0 988887 2007-03-07 2007-06-01 90 3 5.450 205500.00 999.00 999.000 -0.100 2 0 988887 2007-03-08 2007-06-01 90 3 5.450 205500.00 999.00 999.000 -0.100

Attn: someone on R-core: "ran" should be "can". Also, thanks for this feature! Index: Quotes.Rd =================================================================== --- Quotes.Rd (revision 77845) +++ Quotes.Rd (working copy) @@ -74,7 +74,7 @@ Raw character constants are also available using a syntax similar to the one used in C++: \code{r"(...)"} with

Hi, we are a Germany / US based company, looking for experienced Ruby on Rails developer for several projects. The projects ate moduar and planned for a longer timeframe, so we are looking for continous relationship wizth the coders. The jobs is for freelancers/contractors or part-time employees. Andreas Wilkens Please apply by email to jobs-MXf/qEKz1R7qlBn2x/YWAg@public.gmane.org (Attn: