similar to: new application Dialtone()

Hi, is there a way to enable dtmf detection on zap channels? I am trying to pickup, play a ringtone and the dial out. I.e. exten => s,1,Wait,1 exten => s,1,Answer exten => s,2,Playtones(dial) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => _X,1,StopPlaytones exten => _X,2,Dial,Zap/g8/BYEXTENSION|10

I think there are some missing words in "R Internals"'s description of the P_ macro. It currently has "A macro as a wrapper for ngettext", which I think ought to be something like "The macro P_ may be used as a wrapper for ngettext". The following patch also makes the 2 alternate definitions of P_ have the same argument names, StringS and StringP. Expanding the

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

It's not just you. Same thing happens here. I went back to 1.0.7. Stefan Gofferje wrote: > Hi folks, > > I used to have some constructions like > > exten => number/callerid,1,Goto(somewhere) > > After updating to 1.0.8 those does not work any more. > Any hints? > > Regards, > Stefan >

Dear forum, I have a install of asterisk using AMP. I followed the install guide off the AMP site. http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.4.pdf When I start using amportal start or asterisk -ccccv I received this in my log. The last line is that ogg failed. I have found nothing on the web about this, and I am not even sure where to start troubleshooting. Any help

Hi All, I''m pulling my hair out trying to get vpn working from a windows client on the internet to my firewall. I have the following pptpd.conf file: option /etc/ppp/options.poptop stimeout 10 speed 115200 localip 196.22.35.1 remoteip 196.22.35.100-110 196.22.35.1 is my public ip address on my firewall, eth0 ip address, also my lan ip adress. options.poptop looks like this: #ipparam

Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below. The first warning is about music on hold because it

We wrote and submitted a patch to do this. Just modify app_directory.c and recompile. It adds a new flag "b" to the directory( ) app where you can have it use both first and last name. -= Info about application 'Directory' =- [Synopsis] Provide directory of voicemail extensions [Description] Directory(vm-context[|dial-context[|options]]): This application will present

This problem actually came up with pdftex-1.40, which now uses the openbsd-compat code. I routinely build pdftex with c99 to reduce reliance on old gcc extensions, in the hope that things will be more consistent across linux and unix. I think the problem is that the way configure checks for strsep leaves things open for a #def'd version to conflict with the replacement version. I'm

Hi guys, When I try to get ring tones when dialing out with the command Dial(SIP/sipout/${PHONE},15,r), I get the error message indicated in the subject. I've checked my indications.conf file using the sample file provided with asterisk 1.4.10 (the version I'm using) and it's not better. Any idea ? Regards. -- Cyril SCETBON

On my lab systems, the automatic updates were failing because of the problems with ipod libraries from EPEL being in the way. It turns out that was a good thing, because when I "fixed" it, a massive set of packages was updated, including the new gtk3. These packages are the ones causing problems, I think. gtk3-3.22.10-4.el7.x86_64 gtk3-devel-3.22.10-4.el7.x86_64 In the release notes,

I forgot to take out the portion that would read in the voicemail boxes from the text file. If you want to leave it in then you could have some voicemail boxes defined in the text voicemail.conf. I do not, so I have removed it. Below is the new patch: *** app_voicemail.c 2004-06-23 07:55:54.000000000 -0600 --- app_voicemail.c.new 2004-06-23 07:55:47.000000000 -0600 *************** *** 49,61 ****

Hello all, I am just getting going on building my system, but I thought I'd send you all a patch that I wrote so the command: show voicemail users issued from the CLI works properly when there is a postgres backend for the voicemail. The current version of the app does not display the voicemail boxes found in a database. It is called in the load_config function. I haven't done

I have had * installed for a couple of weeks now and am very impressed. I have got Zap, SIP and MGCP working and can call freely between them with just things like transfer still to sort out etc. I then though I would add H.323 support to my working system, having read the previous threads on the subject before I installed I installed the pre-reqs pwlib openh323 gnugk for h.323 gatekeeper

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and

SOFTWARE FreePBX 2.1.3 CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 Sangoma Wanpipe 2.3.4.5 I had an Asterisk server lock up on me today after 95 days of up time. Had to manually kill the Asterisk process and then restart. Nothing out of the ordinary in terms of memory use as far as I could tell. Seems to be running fine now. Here is the log file. I deleted the stuff in the middle to keep it

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

I'm relatively new to Asterisk and I'm hoping attending Cluecon will be a good way to get up to speed on the project and hear about what others are doing with it. We currently use a Cisco IP phone system at my office, although I just added an asterisk box to provide soft phones to our travelling users. (IAX2 is a lot easier to get through firewalls than cisco's protocols). Terry

Installing DevkitLite hardware (Very similar to John Lange's post on Tue Oct 08 2002) I cannot get anything to work on the phone connected to the s100u. I dont know what to do. Can someone please help me? I used the sample configuration files from digium documentaion that was supposed to be "sane" defaults for the kit. Very similar to John Lange's post on Tue Oct 08 2002 Here