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

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about asterisk,I have more did thunk from gateway,how can register?

about asterisk,if I have more did thunk from gateway,how can add? That trunks will register form gateway, and ip is only,but have more did. that register is not sip,is gateway, how can register? i am ...
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21 views

Asterisk hacking defence: “Call setup” in tcpdump from denied IP number

I am running Asterisk 11.7 on Ubuntu for my small business. These last days someone could access the server and make all phone extensions ring in regular intervals. I would like to understand how that ...
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14 views

Asterisk support for rfc 3327 & rfc 6140

Does asterisk support rfc 3327 and rfc 6140 if so which version of asterisk supports these rfc's. Is there any open source sip server registrar that supports the above mentioned rfc's
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21 views

Can't route traffic from Cisco to Asterisk over port 5060

My VOiP provider has informed me that I must route incoming calls from my Cisco router to my Asterisk server to port 5060. I can ping my Asterisk server from my Cisco router fine, there's a direct ...
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32 views

Asterisk server with two connections [closed]

I need that configuartion It works, but with only one connection, with eth0 Or with only mobile broadband I searched for it, but found only forwardind/routing solutions. How i can see asterisk ...
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85 views

SIP.js connection to Asterisk 11.20 over WSS not working

I have successfully setup sip.js using a standard non secure ws:// to an asterisk 11 server using firefox 43. I can make and receive calls to another ff browser/hardphone. But this does not work with ...
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73 views

Elastix 2.5 - Change settings via WEB GUI does NOT take effect

My PBX system is currently running Elastix 2.5, which has been installed & configured about three months ago. For maintaining purpose, I have used etckeeper to tracking changes in /etc folder, ...
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68 views

SIPML5 connection to Asterisk 13 over wss

I have successfully setup sipml5 using a standard non secure ws:// to an asterisk 13 server, can make and receive calls using demo at https://www.doubango.org/sipml5/call.htm. However, I am having ...
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12 views

OPENVPN ipp.txt file assigining duplicate ip addresses

I am having an issue with duplicate entries for hosts in the ipp.txt file where the same host CN has been added multiple times with different ip addresses. For some or other reason some of my openvpn ...
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15 views

Having trouble with one way adio on an asterisk sip phone

I have an asterisk server (Elastix on CentOS 7) that I am currently running in Amazon Web Services. The server works great and clients can connect without issue using a stun server. However I have ...
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20 views

DTMF Tone issue with Elastix

I am very new to Elastix and have just gone through the setup of a new system on Elastix 2.5.0-8, I have Elastix in the middle of 2 sip trunks, one is the service provider and the other is a 3rd Party ...
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23 views

Creating Own DID Number

I have 2 Asterisk Servers installed in different machines. Lets say I have created 3 extensions: 101, 102 and 103 in the first server. Generally we can have a single call connected for one ...
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35 views

asterisk, “blonde” transfer and call back

I have Asterisk 11.x and a Yealink T28P IP-phone. One of my user is requesting "blonde" transfer (this is not my invention, this is the developers term, after all, I did my homework and read a lot) ...
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22 views

What is needed to enable a passwordless SIP connection in Asterisk to accept inbound calls?

I have an Asterisk system with multiple users. Until now, where a user required the ability to place calls, we provided them with a SIP/IAX account with a username and password, plus a custom routing ...
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24 views

How to connect extensions of remote VOIP over remote STUN server?

Image of Architecture I am trying to setup I have configured the Asterisk server in a docker image and it's running, I have also configured the STUN/TURN server and it's also running on the same ...
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63 views

How to connect a client behind a NAT router to an Asterisk server behind a NAT router?

I have an asterisk setup on a server. The local IP address is 172.22.9.200 and extenal IP address is 75.45.185.100. My client device is an android phone that is connected to a router and it which ...
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33 views

Enable Google Authenticator to send codes via SMS with OpenVPN

Setup: Using OpenVPN in single-server<->multi-client configuration. Running google authenticator PAM module in order to enforce 2 factor authentication. Problem: Some employees don't have a ...
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54 views

asterisk Function MAILBOX_EXISTS not registered

I am at a complete loss. I have searched for a while now and I cannot get a mailbox to work. CentOs 6.7 asterisk 1.8.32 freePBX I did this installation through the wiki.freepbx ...
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67 views

Vodafone SIP trunking

I came here to ask if anyone of you has experience with Vodafone Business SIP trunking in Germany or in general. Vodafone offer (german). The reason I ask is because I'm planning on setting up ...
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37 views

NFS, rsync or other to avoid installing ftp on secure remote Asterisk PBX server?

Two VPS hosts: One has everything on it - mail, web, ftp, dropbox and so on. The other is an Asterisk PBX (VOIP server) installation which is locked right down. Nothing except Asterisk runs on it, ...
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36 views

GOIP-1 does not forward SMS messages to Asterisk

I have a GOIP 1 channel gateway and followed this tutorial: https://github.com/tangaza/Tangaza/wiki/Connecting-a-GoIP-GSM-VoIP-gateway-to-an-Asterisk-AGI For me receiving SMS-es only works if the ...
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23 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, ...
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45 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 ...
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152 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. ...
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68 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 ...
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33 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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54 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 ...
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87 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 ...
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25 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 ...
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20 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 ...
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31 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: ...
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609 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 ...
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57 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 ...
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135 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+) ...
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95 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 ...
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69 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 ...
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47 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 ...
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94 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 ...
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11 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 ...
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37 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 ...
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211 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 ...
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1answer
98 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 ...
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135 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 ...
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63 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 ...
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155 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 ...
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56 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. ...
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33 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 ...
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363 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 ...
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62 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 ...
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114 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?