similar to: Returning to Voicemail after returning call

I have an AGI that can sometimes take time complete. I don't want the dialplan to be held up by the agi. Is there any way to call it and have Asterisk continue with the dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190410/4c704231/attachment.html>

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking

Asterisk Users, I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System 2.9.18-4-amd64. A TDM03B is installed on the Debian System. Every time, I try to change my voicemail pin via the Sip phone, the voicemail.conf does not get modify and I see this warning message on the Asterisk command line: [Aug 29 00:12:23] WARNING[19142]: app_voicemail.c:799 vm_change_password:

Group In voicemail.conf I would like to having the following setup per context not per-mailbox settings serveremail userscontext fromstring usedirectory emailbody pagerfromstring dialout sendvoicemail callback review operator volgain nextaftercmd forcename forcegreetings tempgreetwarn Can this be done? Thanks! -------------- next part -------------- An HTML

Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I

If you set "dialout=fromvm" in your voicemail.conf, how do you then go about being able to dial back out? Is there a service feature code?

Using CVS from 7/12/04 and trying to get the voicemail broadcast feature to work. Voicemail.conf has [mycontext] 3722 => 1234,BroadCast Test,,,cc=*@mycontext . then many other voicemail boxes. ----- whenever I leave voicemail at box 3722, only box 3722 gets the voicemail. It is not expanding it to other voicemail boxes in the [mycontext] context. Even if I replace the cc= line with

I current have a working Asterisk 1.2.12 server with ODBC voicemail storage, realtime static maps for voicemail, sip and iax configuration files. Realtime extensions, etc. All works great. I have verified that this configuration works on my test server as well. Now I am trying to test the 1.4B3 version on the same test server, and all works well except for ODBC voicemail. I am using the same

Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute

The advanced voice mail options in RC1 is not providing any functionality except "4 - place outgoing call" and "* - return to mail menu". Below is a sample line from voice mail.conf: 210 => 234,test1,test@dotality.com,,|attach=no|cid=yes|review=yes The is a rew install from the tgz file. No patched have been made to the source code. I did try to apply the advanced10

I am trying to find a way to allow dialout from voicemail when connected from an 'internal' extension context, but prevent dialout when connected from an 'external' extension context. As far as I can tell the dialout context that can be set in voicemail has no regard for the context from which the call to voicemail came in. Any ideas on this? Maybe a variable passed when

Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info.

Hi, I am facing a problem where for legal obligations (LI) I have to copy/mirror/forward the RTP streams for some selected call to an external address/port and I have not found a way to do it with built-in functionality. Do I miss something? The basic requirements are: * Raw RTP (no transcoding, header and payload as is) * Direction (did it arrive at asterisk or was it sent) * End

I have an extension that looks like this: exten => 5555551111,1,Verbose(Door buzzer calling) same => n,Dial(SIP/user1&SIP/user2&SIP/user3) The idea is that any of the three users can answer the phone to let someone in. The problem is that if, say, user2 unplugs his phone then the call immediately goes to his voice mail and the other two do not have the ability to open the door.

say I needed to send a broadcast message that I wanted every user to hear when the pick up thier phone? can I "Play,message" on a line just before they get dialtone? or maybe after they dial before ring? how about a "ringdown" to a voicemail box and on end return them to thier line for the dialout? can * do ringdowns? when a user picks up an extension it automagically

Hello; I am using vicidial which is using asterisk 1.8, mean while when the extension has voicemail, I always see the red light on the Polycom and hear the beep sound (toot toot) in period time. Also, I can see at the LCD an option to select it for accessing the voicemail ?but I am facing the following problems: 1) The red light and the beep: How I can let the Phone only have the red light

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi! Does anyone of you have an idea how to detect an answering machine on a dialout call? I am working an a voicemail system wich calls the subscriber but I don't want to fill their answering machine. Maybe I could detect somehow if there is incoming voice when playing the message. usually real persons don't talk when they listen but answeringmachines do. ;-) Thanks in advance,

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

Hi, I'm just in the process of replacing a crappy Siemens PBX with a new and shiny Asterisk system. To connect Legacy equipment I hooked up a small ISDN PBX (DeTeWe OpenCom 36) to one port on a Junghanns.net quadBRI card. That port is configured for NT Point to Multipoint (Mehrgeraeteanschluss) mode. Now I can place calls from the ISDN PBX to the other Asterisk extensions but the other way