Questions tagged [asterisk]
Asterisk is software that enables a server to act as an IP PBX system, VoIP gateway, conference server, and more.
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Asterisk Cancel Transfer (atxfer)
I'm using the Asterisk 13 AMI to start a atxfer. It works so far.
But how can I cancel the transfer action?
Example:
Bob calls Alice.
Then after some time talking, Alice starts an automated transfer ...
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1
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761
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Asterisk 13 " __set_address_from_contact: Invalid contact uri" error
I'm installing Asterisk 13 on our production Ubuntu 14.0.4 box and the process seemed to go well. I previously installed it on my personal server as well. It works perfectly on my personal server, ...
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Make Asterisk to serve another subnet
I have server with Linux Ubuntu 12.04. It has two subnets - 192.168.0.1(eth0) and 192.168.1.1 (eth1).
There is an Asterisk server (IP PBX) connected to first subnet, lets say it has IP address 192....
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1
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How to have dynamic extensions in Asterisk
I have recently swapped VoIP providers. So far so good, but I have ran into one little snag (and I'm honestly not really sure how to ask / Google this).
With my old provider I could have inside my ...
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How to know what number was dialled on incoming call
The scenario
In the UK.
I have one (BT openreach) phone line coming into a Cisco SPA232D. That phone line is ringable on a number, let's say 01235AAAAAA.
I have a virtual number, 01235BBBBBB, which ...
2
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2
answers
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Fail2ban critical exception - does not block ips
I'm trying to get fail2ban running on CentOS7 (no SELinux) and it uses firewalld. My goal is to set it to ban Asterisk password fails.
Setup was default yum install fail2ban
Config-wise I ONLY added ...
1
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1
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Linux based asterisk server, SIP softphone able to hear audio but not send audio to others
Running Ubuntu 14.04 LTS and Asterisk 11.7.0. The server is set up for SIP phones to use our SIP trunk via voip.ms/didlogic (two different services because we like international rates on didlogic, but ...
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Dynamic Extension creation in Asterisk
I am looking for a way to configure asterisk to allow for dynamic creation of extensions whenever a register command is received. Put security aside for the moment as this is for a development ...
2
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1
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How to match numbers based on end of number?
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 ...
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Make an authenticated SIP call via URI
I'm trying to redirect a phone number to a SIP address that's only accessible when authenticated.
Given the following, how would the SIP URI look like (if possible) ?
SIP Server: sip.example.com
Auth ...
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how do I route Asterisk dials from serverA to serverB?
I'm trying to make an Asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but am having a conceptual problem with ...
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2
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Asterisk call should be with peer, but gets trapped and stays local
I'm working with Asterisk and Vicidial, trying to place outbound calls through a SIP "trunking" provider. The calls never connect, instead I hear either demo-instruct.gsm or invalid.gsm playing from ...
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Asterisk one way audio detection
I have some calls have one way audio. I tried to write ${CHANNEL(rtpqos,audio,all) to CDR and think what if themssrc=0 than no audio, but it's not true. Is any way to detect one way audio?
ssrc=...
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OpenLDAP & Asterisk - no global superior knowledge
I'm struggling to setup OpenLDAP to allow Asterisk to store information about users. I've tried to follow several howto's, but unfortunately I'm not able to set it up correctly.
It seems like most ...
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1
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Asterisk logging to Graylog2
Does anyone have any experience with logging Asterisk servers to Graylog2?
I'd like to be able to send certain verbosity levels to Graylog2, but I don't have a clue how to go about it.
The goal is ...
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3
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The benefits of a proxy server before Asterisk
We are busy to develop an app to let users call eachother using webRTC. On this moment we have audio working, but video not. Now I hear that a proxy server can be the solution for this problem. But it ...
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1
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Asterisk realtime include dialplans
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 => ...
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1
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Asterisk Dialplan Siptrunk dial user input
I have an SIP Trunk on my Asterisk box. So my Asterisk box has one public number and 100 internal numbers.
external number: 0030XXXXXXXX
internal numbers: 2000, 2001, 2002, 2003 ... and so on
I ...
2
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2
answers
3k
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Asterisk WaitForSilence NEVER detects silence
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 ...
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3
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Asterisk new PJSIP driver security option
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....
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why do all softphones fail to authenticate after receiving 401 unauthorized from Asterisk?
When I attempt to connect to the Asterisk server from CSipSimple, over a Cisco router, on the same network, Asterisk logs show the attempted connection:
<------------->
[Mar 23 20:01:34] ...
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1
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FreePBX UK Dial Patterns Route Through Specific Trunk
I've read a ton of links and tutorials and for some reason cannot get my dial patterns to work correctly.
I've got Freepbx installed and working properly - no issue there.
I want to create a dial ...
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Asterisk - very high Load Average with no reason
There is a problem with Asterisk server showing high Load Average I can not explain.
Ny setup is:
Asterisk 13.1.0
Linux 3.13.0-24 (Ubuntu Server)
Dual socket (Xeon E5-2620) server, HT enabled - 24 ...
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Attaching Asterisk console to process
I want to open control console on currently running Asterisk instance. When am I supposed to use option -r and when -c in order to connect to Asterisk process?
What is a main difference?
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WebRTC on standalone asterisk - no audio
After struggling with Asterisk for WebRTC for a few weeks now, I decided to put my problem on this forum. My Problem is as follows:
Im not getting audio from WebRTC to WebRTC clients. I work in a LAN ...
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2
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Can't dial through SIP trunk: FreePBX/Asterisk
I set up a SIP TRUNK in FreePBX/Asterisk that works perfectly for incoming calls. Here' s the relevant configuration:
type=friend
host=201.217.31.10
callerid=mynumber
username=595XXYYZZZZZZ@prepago....
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1
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2k
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Asterisk Sip Server and "Screen Sharing" function
I'am newbie in asterisk, i followed the tutorials on the asterisk wiki, and installed the latest version (13) of the asterisk server. I was able to setup the voip now i wonder, there is a function im ...
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Asterisk: execute shell command on server when call is accepted on SIP extension
I'm trying to configure asterisk to execute shell command for incoming calls - but only when the call is accepted. I've managed to setup extensions.conf so that command is executed when new call comes ...
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Asterisk behind NAT sets wrong Contact Header
I'm using SIP with asterisk 13.1.0 behind a statically configured NAT.
The servers private_ip differs from the public_ip, where I can reach it.
I've already set these options in the sip.conf file.
...
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Asterisk - Send voicemail to email as client SMTP address
I'm able to setup Elastix (Asterisk) to send email voicemail attachments without any issue. But I'm trying to email them to a particular external address and it seems to be rejecting the emails as ...
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591
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Using Asterisk for IVR calls without internet connectivity
I am thinking about using Asterisk as an IVR for a project in Burundi, in a place where we often lose internet connection. I'd like the Asterisk server to deliver IVR calls through, say, an Android ...
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755
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Asterisks Dialplan Extensions
I'm playing around with a very simple asterisks setup. My asterisks server is 192.168.1.10 (Ubuntu 12.04), my 2 sip clients are 192.168.1.20 and .21 (both using ubuntu 12.04 and Jitsi as a sip client)....
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sip register success on one host but fail on another host in same local network
I acquired antisip account and setup asterisk (lets say PBX-A) to register. It worked fine.
However, I attempted to do so with another asterisk (PBX-B). No luck this time.
PBX-A and PBX-B are on the ...
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2
answers
798
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routing for iax-protocol does not work on pfsense
I have problems with a pfsense-box routing iax-traffic (UDP port 4569). Our company has a central office and eight branch-offices. At the branch-offices we use pfsense as firewall and VPN endpoints (...
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Finish extension (avoid dialplan to silently continue in the next priority of another extension)
I have run into a strange behaviour:
[ctx-test]
exten => h,1,NoOp(*** hangup ***)
exten => _.,1,NoOp(*** ${EXTEN} ***)
same => n,...
same => n,...
same => n,Hangup()
Strangely, ...
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2
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4k
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Asterisk Firewall access Rules
I have a questions regarding the firewall rules for Asterisk VoIP server.
I have a restricted access to UDP/TCP port 5060, which seems to be blocking calls.
I want to make sure that my VoIP Server ...
2
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1
answer
800
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Log to file in Asterisk: append with newline
(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 ...
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1
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649
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Asterisk Event PauseQueueMember - AMI Calls
We use Asterisk-AMI and we have a little JS server which listens for Events from the AMI to gather & display statistics (Calls, Queues etc).
I initialy thought that when a queue member is paused, ...
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3
answers
3k
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configuring command prompt in Asterisk CLI
I am trying to set up color prompt in asterisk CLI. In the documentation I have found this:
%Cn[;n] Change terminal foreground (and optional background) color to
specified. A full list of colors may ...
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1
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518
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Asterisk and FXO issue over DDNS
I've been experimenting with asteriskNow (freePBX/asterisk 11) as a sip server with DDNS. Everything seems to be working fine using externhost and such; calls to remote extensions to and from asterisk ...
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Call duration limit & least cost routing
We have Elastix version 2.4 setup and we have some little questions which if you could please answer to solve via web based panel of Elastix, as agents are making calls via Elastix there are little ...
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1
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Asterisk can not send fax via T.38
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 ...
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0
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Asterisk 11.* TLS Configuration
I have installed asterisk 11.12.0 on CentOS (64-bit). When enabling the TLS support i get the following error
WARNING[7620]: tcptls.c:673 handle_tcptls_connection: FILE * open failed!
i tried ...
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Asterisk: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
Strange issue started with our Asterisk install yesterday.
When dialling out via ISDN BRI the call fails and the following error is logged:
Unable to create channel of type 'DAHDI' (cause 34 - ...
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3
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3k
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Asterisk ring out even when the client is offline
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-...
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2
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Asterisk: Join 2 inbound calls
I have a VoIP box setup with Asterisk and using chan_dongle to provide me with an inbound GSM trunk as well as a couple of DID SIP trunks with local numbers. I would like to be able to have the ...
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539
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Using IAX2 to call from one Asterisk server to another
I have two asterisk servers serverA and serverB . I am trying to place a call from serverA to use serverB using IAX2. I have defined a user on serverB in iax.conf. When making the call this is what I ...
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1
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How to know if graceful restart / stop was executed?
Is there a way to know if restart gracefully or stop gracefully has been executed on asterisk while still running and waiting for calls to end then, restart/stop?
When this mode is executed, Asterisk ...
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1
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1k
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Asterisk relay between multiple subnets
I wonder what's the best way to go when you have phones on multiple networks which are not directly reachable. I have 3 networks
10.3.x.x
10.6.x.x
10.17.x.x
My asterisk server resides on the 10.3.0....
0
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1
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181
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Asterisk test ISDN card
I have an AVM ISDN controller C4 plugged into an asterisk machine.
It shows up in the PCI device list as:
Digital Equipment Corporation StrongARM DC21285 (rev 04)
How can I test this card to see ...