Asterisk is a communications server software and an open source framework for building communications applications.

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8 views

Execute dialplan or a script once Asterisk started

Is there a way that I can execute something inside Asterisk or maybe a script once Asterisk has been started? I am planning to update a record in db table once asterisk started (like resetting ...
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14 views

Billing for inbound calls using A2billing and Asterisk

I have installed A2billing and integrated it with Asterisk, Is there a way to test A2billing for inbound calls ( I mean asterisk to asterisk or local users)? I just want to add define rates for ...
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1answer
24 views

unicode characters suddenly appearing in asterisk CLI SSH console

I have a mysterious issue. I connect to an Arch Linux virtual machine running asterisk. All of a sudden, anytime I enter the asterisk CLI, the SSH console spits out unicode character codes. What I am ...
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1answer
17 views

is this failed SIP call due to firewall or NAT?

I'm trying to find out why this call failed, with: <--- SIP read from UDP:63.247.69.226:5060 ---> SIP/2.0 100 Trying and then Everyone is busy/congested at this time (1:0/0/1) full debug ...
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23 views

How do I troubleshoot a Cisco IP phone with no sound? [on hold]

In provisioning the Linksys branded Cisco IP phone described below I think there's an error with regards to the dialplan in the hard phone, or perhaps ports, preventing sound. Admittedly, I have no ...
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29 views

Why doesn't nmap show SIP ports open for this router?

SIP calls on from demo_bob to demo_alice on Asterisk had no sound, so I opened ports on the NetGear WNR1000v2 - N150 Wireless Router: # Service Type Port IP 1 IP_Phone 6670-6670 192....
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24 views

How is a SIP invite sent?

The telnyx Data API tab describes a SIP request: Requires IP Authentication to be setup through the portal and associated with LRN under Telephone Data Tab Send a SIP Invite to lrnlookup....
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1answer
26 views

Does Asterisk support Group Chat?

I am fairly new to Asterisk, I have setup a server with ability to call and instance messaging, I have done some research about adding Group chat to asterisk but no result yet. Does Asterisk support ...
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3answers
56 views

how do I provision a hard phone for getonsip?

In provisioning a Cisco SPA 942 hardphone, branded Linksys, how do I finish the setup for getonsip (or onsip)? SIP Address: foo@getonsip.com Username: foo Domain: getonsip.com SIP Password: ...
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17 views

SIP server name Defaults to $VERSION for Cisco SPA 942 hard phone

In reading the fine manual for a Cisco SPA 942 IP phone, branded as Linksys: what is, in particular, meant by: SIP Server Name Server header used in responses to inbound responses. Defaults to $...
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27 views

PJSIP channel between asterisk servers

Currently I stuck with the configuration of a channel/trunk with PJSIP in Asterisk (version 13). What I want to do is the following: I have two, three or more asterisk servers on different sites ...
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1answer
18 views

Asterisk - DAY & NIGHT

I slowly desperate yet. I simply do not understand the principle of include in extensions.conf. i hahe this code in my extensions.conf: [GLOBAL] include => technik|15:00-17:00|mon-sa|*|* include =...
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1answer
56 views

Installing Asterisk extra sounds on Fedora 23

On Fedora 23 I found that the Asterisk "extra" sounds don't seem to be included in the main DNF repos. How can I install the "extra" sounds using DNF? For example, I would like to be able to do: dnf ...
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1answer
52 views

Asterisk connectivity issues

So I am having rather an odd issue that I do not know how to investigate. I installed Asterisk with FreeBPX on a raspberry pi and configured it to use IAX, used a softphone to connect and everything ...
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1answer
59 views

Play music on Asterisk without blocking extension

I'm writing an Asterisk dialplan using Lua. First, I'm reading a sequence of numbers then I send a HTTP Post to a webservice. I would like to play some music to the user while the script performs the ...
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1answer
70 views

ISP blocking 5060

Here is a problem. I move my RasPBX Server (Asterisk for Raspberry Pi) to new apartment and ONLY one ISP in building which is blocking 5060 After few days fighting with Asterisk, I discovered that ...
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34 views

Asterisk Win32 (SIP) Spawning Thousands of Procceses When Run

A bit of background: Recently I had found Asterisk for Windows and decided to see if I could get it set-up on a small home network. I downloaded the install file from the download link. Next I ran ...
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2answers
168 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 has ...
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1answer
105 views

asterisk 11.7 with app_mp4 for playing mp4 videos

I have successfully compiled and run app_mp4 for asterisk 11.7 and recording audio/video with MP4Save. I am using ulaw as audio codec and h263 (simple h263 not 1998 or 2000) video codec. Although I ...
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2answers
341 views

mp4 video for asterisk with ulaw as audio and h263 for video

Scenario: I want to play a video file on asterisk. my softphone only support ulaw/h263 for audio and video respectively. So I am allowing both formats in sip.conf file and expecting it to run. ...
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1answer
215 views

how to make Asterisk Reject SIP call with code 503

I'm writing a dial plan using phpAgi, there are some cases that it should reject an SIP call with error code 503 (service not available) I'd appreciate your help
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2answers
52 views

Asterisk and x100p - Error: wcfxo: Out of space to write register 05 with 0a

I am running a IP PBX system with trixbox, what i'm trying to do is connect PSTN lines with IAX2 using X100P analogue card. The problem is i cannot call any outgoing routes. (already tried ...
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7 views

Asterisk+freepbx and a2billing [duplicate]

I have an asteriskNOW up and running, soon will connect it with ISDN to make PSTN calls, now my users need to call through IAX2 to PSTN. So how do i create balance account for every single person?
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1answer
119 views

How to create account balance for each users in asterisk freePBX system? [closed]

I have an asterisk server up and running. Users connect via extensions(IAX2). How do i create balance account for these users? any help would be appreciated.
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2answers
1k views

How do i run asterisk behind an VPN?

I am new to asterisk, so basically i know nothing about IP PBX. I need to start the a VPN server with behind it asterisk and my clients should be able to connect with softphone with build-in VPN ...
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147 views

Sending special characters sms from asterisk CLI

I've setted up an asterisk with a gsm card (openvox a400p) succesfully. I've tested send sms from CLI and it works great. The problem is with special characters: CLI doesn't allow me tu put any ...
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50 views

Confbridge - Set speaker's volume just for a specific listener

I use Asterisk to simulate air trafic communications (FGCom for the FlightGear project) and I'd like to simulate the signal weakening with the distance. To do that I thought to send from clients to ...
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31 views

Setting up landline on mobile using Asterisk

I recently got a new broadband connection with a dlink DSL-2730U N150 wifi router. I also got a landline connection through the same line connected to my landline phone. I read at places the I can ...
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1answer
471 views

Asterisk SIP calls ending after 2 minutes using Twilio's trunking service

I have an Asterisk server hosted on DigitalOcean that is having calls drop after exactly 120 seconds using Twilio's trunking service. It is notable that directly connected softphones do not drop their ...
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1answer
817 views

How to call a PSTN number from a soft SIP Phone through Asterisk

I am getting started with Asterisk. All I want to do is to call a PSTN number from a SIP softphone. I have gone through the latest edition of the book: Asterisk, the Definitive Guide and done the ...
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1answer
233 views

Make one way SIP video call

Using Asterisk, CSipSimple and MicroSIP I was able to make a video call, but to make it work both parties should click on "video" and video will be transferred in both directions. I do not like this. ...
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1answer
64 views

Do I need STUN?

My router has a "real" static outer IP address. Behind router's NAT there is an Asterisk server. Router is configured to forward 5060 UDP port to Asterisk machine. Do I need STUN server assuming that ...
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355 views

SIP: The other party can't hear me but I can hear them when using Asterisk

Please read carefully all the question and do not rush to a conclusion. === Test environment for point-to-point calls, without Asterisk === City1: A router with NAT and WiFi and 5060 UDP port ...
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1answer
102 views

What does Asterisk need codecs for?

Let's take a system which consists of one Asterisk server and two SIP clients. Each client has an equal set of codecs installed. Traffic encoding and decoding is performed on a client side. Then why ...
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2answers
2k views

Failed to authenticate SIP peer

I am trying to connect an SIP peer using Zoiper to my asterisk server. The peer is a soft-phone on my server. I have the following config for the peer: [201] disallow=all allow=alaw host=192.168.1....
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1answer
48 views

What's Lax and Lax2 protocols?

Are LAX and LAX2 communication protocols like SIP? What are these? Some telephone systems support them beside SIP but I cannot find information on google
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1answer
476 views

How do I run Asterisk as user other than root?

I compiled Asterisk 11 from source as user root on CentOS. Now I want Asterisk to run as user asterisk group asterisk. I have uncommented AST_USER="asterisk" AST_GROUP="asterisk" in /etc/sysconfig/...
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62 views

Ekiga softphone shows as “registered” — but is it actually?

Is Ekiga actually registered? How would I know? I think Ekiga connects, or at least it says "registered": But, when I exit Ekiga, and use SFLphone, it fails to register: While sipsak shows a ...
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186 views

Error: “Channel not in Stasis application” when adding channel to a bridge

I'm working with Asterisk ARI. When I create channels via ARI, I specify application name and I can add those channels to a bridge with no problem. But when I use other client (like Zoiper) to make ...
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1answer
93 views

Asterisk / Digium card / FreePBX : how to define on which inbound port I want to connect the telephony line on my 4-ports telephony card?

I am a newbie with Asterisk. I have a 4-ports Digium telephony card. I connected my analogic telephony line to one of these 4 ports. I have set up an IVR thanks to the FreePBX interface. I have set ...
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1answer
157 views

Not registering on Asterisk over VPN

I want to work in local area from home, I configured OpenVPN on my Asterisk server (it is only one Linux server in LAN), it works fine. I enabled forwarding on Asterisk server, and client now can ...
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1answer
1k views

VOIP Fax + Raspberry Pi

I just moved into a new office and don't want to get a landline phone. I'd kind of like to set up an Asterisk box and use Google Voice as my service provider. The main things I'd like to have are: ...
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1answer
346 views

Asterisk Ivr menu mysql build

I would like to implement an IVR menu to see if the call comes in and announces a text waiting 6 numbers. Then the number 6 is controlled by a MySQL database. If "Active" or "Valid" the status will be ...
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1answer
360 views

Getting Asterisk PBX to do SIP URI dialling properly?

I've installed AsteriskNOW in a VM and I'm having a hard time getting calls from the PBX to head outbound to another SIP address on the SIP2SIP network I use for occasional testing purposes. However, ...
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768 views

Ubuntu + Asterisk = Registration from '“100” failed for '10.0.83.202:51966' - Wrong password

I have set up a simple Ubuntu 14.04 just installing Asterisk (apt-get update && apt-get upgrade && apt-get install asterisk). When I try to log in from my client (Telephone for OS X) ...
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1answer
306 views

Can Asterisk accept and send SMS?

Is it possible to send and receive SMS over Asterisk? I am using Asterisk for my communications and it works pretty good. I just wish that I could send SMS to my callers. thanks
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1answer
166 views

Asterisk : how to connect my PC with Asterisk to the phone line?

I am completely new with Asterisk. I have installed Asterisk on my PC. Now, I want to connect my PC to the phone line. How can I do ? To be more accurate : how can I do if I use an internet box ...
2
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0answers
229 views

How to implement an integration for start and end of a call in Vicidial

I've developed most of the application in PHP for the integration (very similar to our Asterisk implementation) and all we need at this point is to run php async.php 12313123.12312 (shell command) (...
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129 views

Digium card shows green and red light after each second after configuration

My digium card and pri was working fine in other system but i removed and installed that digium card in other system .I have configured it but its showing only 24 channels but i have 30 channels .My ...
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1answer
5k views

kernel source error on dahdi installation in centos

Trying to install dahdi from the link. I have downloaded package for dahdi. But on doing cd /usr/src/dahdi-linux-complete* make && make install && make config My terminal shows [...