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

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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 ...
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2answers
135 views

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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2answers
2k views

Asterisk IVR setup: disallow direct dial for some extensions?

We've got PBX to configure. It's based on Elastix 2.4.0, freePBX 2.8.1 and Asterisk 1.8.20.0. Currently we have a setup for inbound calls like that: Inbound call > Announcement (30 sec) > IVR ...
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0answers
7 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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2answers
19 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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1answer
6k views

Asterisk cached an old dial plan? rejected because extension not found in context?

I'm trying to fix my asterisk server which has been quite stable up until recently. Here's my problem: a) if i run a sip reload and/or dialplan reload from asterisk terminal, it sometimes breaks the ...
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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 ...
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1answer
87 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 ...
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1answer
2k views

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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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 ...
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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 ...
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1answer
1k views

Asterisk AGI randomly returning 4 and failing to execute PHP AGI

I've got a PHP script that I execute 2-3 times throughout a single calls life time, if it is inbound it will execute once for the queue, once for start of call and a last time for end of call. For ...
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1answer
2k views

Unable to connect to Asterisk via Telnet

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 ...
5
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4answers
9k views

How to configure Asterisk to send audio before call is established

Is it possible to configure Asterisk so that it sends RTP packets with audio from the receiver before the remote party actually picks up? This seems to be required for a VoIP compliance test my ...
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2answers
1k views

Asterisk DID incoming rules =s

So I have a 10 way SIP trunk from a UK provider. Calls in and out work fine, but as I have multiple numbers for different purposes I set up the inbound rules for each number but they never work, only ...
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0answers
22 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
1k views

Asterisk Real time does not perform register with SIP TRUNK provider

I am using astersik real time (dynamic). I have entered my SIP trunk details into the sippeers tables. However, the sip trunk does not perform a register with the SIP TRUNK providers servers as it ...
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1answer
1k views

Accept calls on external cell phones - FreePBX Distro

We installed FreePBX distro last week and configured it so that when someone calls in, it goes to all the phones so someone here can answer the call. After trying all the internal extensions, it ...
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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 ...
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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 ...
2
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1answer
1k views

asterisk/freeswitch in nat/no-nat setup

my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party ...
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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 ...
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2answers
1k views

elastix cdr stop working

CDR was working before 19 march. Unfortunately i dont remember what kind of changes i made to configuration, but this exactly not changes to CDR config. elastix 2.4.0 asterisk 11.7.0 mysql 5.0.95 ...
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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. ...
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0answers
30 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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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 ...
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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 ...
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2answers
1k views

Make call with alternate provider if NOANSWER

I have two voip providers, one free an the other paid. The free provider only allows local calls to certain area codes, so I need to fall back to the the paid provider if a call fails. At the moment, ...
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0answers
148 views

Asterisk GOIP Callerid not working

Have anyone made this work with GOIP or other similar voip gateways with asterisk? I think I have exhausted all possible options regarding setting a caller ID and it still doesn't work. The recipients ...
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1answer
111 views

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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1answer
64 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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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?
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1answer
94 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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1answer
170 views

secAst lib prerequisites too new for Debian/Utuntu 12 LTS?

as requested in FAQ I wrote here the question (there is no question for this product yet on this site?) This Asterisk securing tool looks very nice so I tried to test it out: ...
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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 ...
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2answers
1k views

FreePBX How to play an announcement for misdialled calls

Our Asterisk-based PBX (FreePBX "Tango" distro) does not require any prefix for outgoing calls. The system is able to recognise external numbers and route them appropriately. Some of our users who ...
2
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2answers
229 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 ...
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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: ...
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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 ...
3
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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 ...
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1answer
8k views

Asterisk and SIP behind NAT

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 ...
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0answers
170 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 ...
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1answer
84 views

Asterisk (Trixbox CE) sound to low in message2email

Hy in Asterisk (Trixbox CE) latest version, we receive message to our email in WAV. But the sound it's too low. Where can I fix that?
2
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1answer
471 views

Sending faxes from Asterisk (email to fax) — is any feedback possible?

I am about to make some email to fax gateway --- some python program will make Asterisk call files and put them into some queue (files or db based). Then another program should take "call tasks" from ...
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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, ...
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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 ...
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1answer
116 views

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 ...
0
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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 ...
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0answers
197 views

Connect Cisco 7975 VOIP phone to two separate Asterisk PBX's

I have a Cisco 7975 registered on an Asterisk SIP server, and I want to register a second line on a second (different IP) Asterisk SIP server. I tried to do this with firmware version 8.5.3 but it ...
0
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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 ...