I am setting up an Asterisk VoIP server (using FreePBX) and I need to identify all numbers that end with a particular pattern.
This is required for routing purposes, that is, to find out whether they come from an outbound trunk or an internal phone.…
I am trying to use my dialplan to play recordings with WaitForSilence to make sure it wait until the person is done speaking or the message is left on voicemail. However, it doesn't seem to wait for 5 seconds of silence. Even if I'm talking it will…
I noticed that when creating a dialplan in asterisk on realtime, I cannot inlcude contexts.
So to be clear, I want to do the following:
[context1]
switch => Realtime
[context2]
include => context1
switch => Realtime
Or
[context1]
switch =>…
I have a working asterisk environment, but I get a lot of unwanted traffic, like sip scanners of people who even try to call as a guest.
I'm using res_pjsip, the configuration is stored in pjsip.conf. But I can't find options like alwaysauthreject…
(I am testing on Asterisk 11.7.0~dfsg-1ubuntu1)
I am using the following format to append to a logfile, according to the documentation:
same => n,Set(FILE(/tmp/mylog.txt,,,a)=my-log-message)
But this does not append a newline. So I am trying:
same…
I am a newer on asterisk. when I tried to send fax, I can not get success.
After doing a lot of reseach, I decide to ask my question at here.
I hope someone could give me some idea to correct my configuration. finally, I want my system can send fax…
I need to configure an Asterisk box to go to voicemail but only after X ringouts.
exten => 1234,1,Dial(SIP/ivan, 30)
exten => 1234,2,VoiceMail(777@mb_tutorial)
exten => 1234,3,PlayBack(vm-goodbye)
exten => 1234,4,HangUp()
The client in my case is a…
I have SecAst up and running great, and I realize that any workstation on the network can access the SecAst telnet interface. Is there a way to limit this to only my laptop? I don't want one of the end users to mess with the ban/unban interace
We are implementing Asterisk 11.4 and using the native PostgreSQL database to store call data. So far, we have not been able to change the admin password through the PBX manager.
The password gets changed in the manager.conf file. However, the PBX…
I'm trying to connect to asterisk via telnet. In order for PHPAGI to use certain commands, it has to be able to do the same. So I'm cutting out the middle man and testing it manually. I've determined that asterisk is not letting anything connect…
I have asterisk-Freepbx (Version 12) hosted on a debian 7 server. I have created an extension (Cisco IP phone SPA 504G). I am able to dial in and out. The extension is configured to go to voicemail if unanswered, busy or unavailable. However, if…
We have people in our office that use softphones to connect to our Asterisk system. The softphones are online when they are in the office, and offline when they are not. So the Reachable/UNREACHABLE notices in the log are an accurate enough…
I am trying to Setup an Asterisk-Server to accept calls from a client in an other Network. The Server and the client are behind an NAT.
I have already activated STUN on the client, but I am still having problems hearing the other side on both.…
I am trying to pass call via sip phone and I was success when I tried for the first time but after that I can't connect the call showing the message below. Can any one give me a solution why this happening, if the configuration is not ok then how I…
We have several Mitel 5224 SIP phones, but can't seem to find out to make the ringer volume setting persist after it reboots. Can the ringer volume can be set somewhere in the XML configuration file?