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

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2
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1answer
31 views

No application 'Push'

According to https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PUSH, the command Push() should be perfectly valid. However, in my Asterisk 13.6 installation: same => n, Push(...
2
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2answers
82 views

Asterisk PBX - What connection to use: wireless or ethernet cable

For a firm with about 8-10 employees what connection should be good enough to connect with Asterisk server. Right now we are all connected via wireless router, but if call quality would be better we ...
4
votes
1answer
858 views

High availability Asterisk options?

We currently run FreePBX on a single node. I'd like to have a cluster of asterisk instances for sharing the load, but mainly for failover. I'm curious how other people have solved this problem. ...
1
vote
1answer
145 views

Asterisk with softphone and webphone

I have a problem with Asterisk. I wrote a WebPhone, so I should add to my peers some config details: encryption = yes avpf = yes icesupport = yes dtlsenable =yes dtlsverify = no dtlssetup = actpass ...
0
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1answer
64 views

Dynamically assign extension based on active disrectory SID with asterisk

I'm new to Asterisk. Basically all I need to do is set up Asterisk to have dynamic extension numbers based on Active Directory (LDAP) SID. AKAIK the SID is always the same prefix, followed by unique ...
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0answers
117 views

Configure Elastix to show incoming caller name from predefined list

I have configured call center in Elastix. My task is to show caller name from predefined list of contacts when incoming call starts. Right now I have only numbers under Source column in Monitoring ...
-1
votes
3answers
95 views

Setting up 2 NICs, 1 for Internet 1 for Phones

I have FreePBX setup and it has 2 NICs. I've been trying to configure it out so that: eth0 will continue to be the one used for all the office phones and internal calls; and eth1 is the new NIC ...
0
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2answers
29 views

Asterisk Transfer call to next extension if previous INUSE

I am trying to transfer call to next extension if previous is using (INUSE) or call is in progress. I tried to use EXTENSION_STATE(extension[@context]) to find the status as follow: [sales] exten =&...
1
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0answers
25 views

Voicemail location with two Asterisk servers connected by Follow-me

Here's our current setup: We have two Asterisk servers in different buildings setup and connected by IAX. Call them ServerA and ServerB. We have several DID trunks coming into Server A, where our ...
0
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2answers
32 views

EC2 Server Seems to have port issues

I have installed an Elastix server on a m4.large instance,but I cannot get sound on any of my calls. The server is configured as follows: CentOS 5.9 Elastix 2.4 Using the following guide: https://...
1
vote
1answer
2k views

How to configure a custom context/dial plan for incomming calls in Elastix/FreePBX?

I would like to know how set a custom context from the extensions.conf file, to deal with the incoming calls in Elastix/FreePBX? I have developed a custom dial plan called [my-custom-incoming1] and ...
0
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1answer
83 views

Softphone not connecting Asterisk

I am trying to connect my X-Lite with a remote Asterisk server. I had installed that on a Ubuntu VPS server (14.04) following this tutorial: https://www.callcentric.com/support/device/asterisk/1_8 I ...
0
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0answers
218 views

Asterisk recieves 403 on every outbound call after 5 minutes delay

I have a home server in my local network with Asterisk installed. It has been working without any problem for several month. I have just changed my router (ZyXel Keenetic Lite -> LP-LINK TL-R480T+) ...
-1
votes
3answers
140 views

Centos server provides javascript for localhost but not local lan [closed]

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
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0answers
128 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
2answers
82 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
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2answers
201 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
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0answers
17 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
48 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
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0answers
336 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
1answer
199 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
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0answers
215 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
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0answers
118 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
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0answers
278 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
75 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
39 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
864 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
96 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
171 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
766 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 ...
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0answers
82 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 ...
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0answers
600 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
1answer
437 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
240 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
337 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 192....
0
votes
1answer
213 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
240 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
436 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
792 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 ...
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votes
1answer
74 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
150 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
221 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
81 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
2answers
570 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 ...
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0answers
98 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? ssrc=...
1
vote
1answer
816 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
322 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
819 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
520 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
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1answer
135 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 ...