Asterisk is software that enables a server to act as an IP PBX system, VoIP gateway, conference server, and more.

learn more… | top users | synonyms

0
votes
3answers
37 views

Centos server provides javascript for localhost but not local lan

I have a centos 6 server running an asterisk, freepbx, and apache 2.2.25. On this server, the apache server serves up a webpage called fop2. I've talked to the developer, and he says the files are all ...
0
votes
0answers
8 views

Calls on new telephony Server (Vicidial) being “dropped” after 30 minutes being INCALL?

Hello everybody! Im in a weird situation with my Vicidial Cluster with the SIP Account 1. I have actually 3 Servers: 192.168.1.200 (All-In-One Server) 192.168.1.201 (Telephony 1) 192.168.1.202 ...
1
vote
1answer
13 views

Asterisk write cdr now

I'm writing CDR to mssql table with cdr_adaptive_odbc. This works fine in the sense that the record is written after the call. But I'd like to write a cdr_record when the call comes in, before it is ...
0
votes
2answers
20 views

Does a SIP trunk relay the RTP stream or just set up the call?

Suppose I have a SIP PBX like Asterisk and a bunch of phones registered to it, and outgoing/incoming calls are handled through a SIP trunk. Do the RTP streams go directly between the phones and the ...
0
votes
0answers
6 views

How to recognize conference calls from individual simultaneous calls When we have Asterisk user agents?

I want to know how I can distinguish between conference call from some independent simultaneous calls when we have Asterisk agents? RFC 4353 says that in conference calls we have a call member with a ...
1
vote
0answers
12 views

Asterisk redirect incoming call to last number that called it

I would like to know how to redirect an incoming call to the extension that called it first in Asterisk. I have not found a way to do it, and I haven't found any Asterisk variable that contain a value ...
0
votes
0answers
23 views

Asterisk SMS with GOIP

I have a GOIP 1 channel gateway and following this guys tuorial: https://github.com/tangaza/Tangaza/wiki/Connecting-a-GoIP-GSM-VoIP-gateway-to-an-Asterisk-AGI For me receiving SMS-es only works if ...
0
votes
0answers
32 views

SecAst Web Interface

Im running an asterisk 13 server with a freepbx frontend and I've be trying to get SecAst working properly. I've installed and configured it and it's running without errors. I can telnet into it ok ...
0
votes
0answers
33 views

freeswitch call getting abandoned

i have a freeswitch box that successfully handles incoming calls to a certain DID number, but when i call in from another DID number (which happens to be from Asterisk) i get a message that the call ...
0
votes
0answers
17 views

OpenSSL conversion from Windows CA for Asterik / Switchvox X.509 PEM Format

I'm trying to sign our switchvox PBX web certificate with our windows AD CA. SV Keeps telling me the certificate is invalid and must be in PEM format. Can anyone see what i am doing wrong? I Think ...
0
votes
0answers
31 views

Asterisk SIP Retransmission timeout after a month of server uptime

This Asterisk server runs fine behind NAT with a dozen of SIP registrations and the call volume of a few hundred calls per month. But then, after about a month of uptime, out of the blue the SIP ...
1
vote
1answer
34 views

RPM built on same Guest OS, segfaults on another host (Asterisk)

I have built an RPM for Asterisk 13 for Centos 7 using a build VM running on OSX (MacBook, using Parallels). This is with the intention of using this RPM on both our QA and production systems. ...
0
votes
1answer
25 views

Will a 12.28“ x 4.2” x .68" PCIe card fit in a Dell R510?

I am planning for a phone server and plan to get a Dell R510 to house two Asterisk 2400 series PCIE cards. (model 1AEX2406EF to be exact). I know the card is 12.28" x 4.2" x .68" (31.19 x 10.67 x 1.72 ...
0
votes
1answer
65 views

Asterisk does not use externip

I am trying to setup a cloud Asterisk server that is behind a NAT with the hello-world example. I have NAT issues. The IP address Asterisk is supplying to the client through the SDP is its local ...
0
votes
2answers
36 views

Asterisk dialplan several apps simultaneously

I am designing an Asterisk 11 Dialplan for a call center. I am facing a problem in dialplan. In order to achieve the desired configuration , i think it is necessary to run 2 applications at the same ...
-4
votes
1answer
56 views

Can I use Kamailio Sip Server to handle 500 calling? [closed]

I need a Sip server to build a VIOP server to establish voice call between 500 Android an desktop sip clients. Can Kamailio handle this or I need an Asterisk server too?
0
votes
1answer
96 views

Can't get voice after changing Asterisk UDP port

Test environment consists of a CSipSimple client, Asterisk server and a router with NAT. Formerly the equipment used UDP port 5060 and everything was good. Then I changed port number to 15000 in ...
0
votes
0answers
66 views

Asterisk SMS to Email (The best way to do this)

I have a Goip voip gateway connected to an Asterisk box. What I would like to do is email all incoming SMS-es to a single defined email address. The problem is that this gw only functions the way ...
0
votes
0answers
46 views

Any solution for ODBC disconnections in Asterisk?

I have configured my Asterisk real time using ODBC. I am using MySQL as data base. I am getting these below messages on Asterisk CLI frequently and immediately my server console is disconnecting: ...
1
vote
0answers
24 views

Issue with Asterisk on Google compute engine (GCE)

I installed and configured Asterisk in Google compute engine (GCE). I am facing the below issues: 1) Unable to login to Asterisk CLI by using asterisk -rvvvvv command (But, I am able to login using ...
1
vote
0answers
171 views

Asterisk dialog destruction on attended transfer timeout

Steps to replicate: Alice receives incoming call from Eve (either from external line or another extension, doesn't matter). Alice makes attended transfer to Bob and hangs up before Bob answers. Bob ...
0
votes
0answers
87 views

Asterisk : SIP/2.0 401 Unauthorized : same phone , different results

First of all I have already taken a look at similar questions such as : http://serverfault.com/questions/392979/asterisk-sip-2-0-401-unauthorized or Asterisk 401 Unauthorized when trying to register ...
0
votes
1answer
89 views

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 ...
0
votes
1answer
104 views

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, ...
0
votes
1answer
147 views

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 ...
0
votes
1answer
91 views

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 ...
1
vote
0answers
94 views

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
votes
2answers
230 views

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 ...
0
votes
1answer
346 views

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 ...
-1
votes
1answer
51 views

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 ...
1
vote
1answer
106 views

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 ...
0
votes
1answer
41 views

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 ...
0
votes
1answer
47 views

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 ...
0
votes
0answers
97 views

Asterisk Auto dial out issues

I'm trying to make auto dial out in asterisk. I have created call file in /tmp directory and moved the file to var/spool/asterisk/outgoing/. It's not working. Please help me to resolve this issue. ...
0
votes
2answers
198 views

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 ...
1
vote
0answers
52 views

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? ...
1
vote
1answer
304 views

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 ...
1
vote
1answer
116 views

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 ...
0
votes
3answers
174 views

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 ...
0
votes
1answer
148 views

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 => ...
0
votes
1answer
53 views

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 ...
0
votes
2answers
188 views

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 ...
0
votes
2answers
128 views

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 ...
1
vote
1answer
121 views

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] ...
0
votes
1answer
234 views

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 ...
3
votes
1answer
344 views

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 ...
0
votes
1answer
17 views

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?
0
votes
0answers
102 views

Asterisk using PORT NUMBER in FROM FIELD of SIP calls

I have an annoying situation with Asterisk 13. I setup a SIP trunk that works OK to receive calls. However, I can't make calls even though my SIP provider is working (I can place calls with a regular ...
4
votes
0answers
559 views

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 ...
0
votes
1answer
634 views

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 ...