Questions tagged [asterisk]

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

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16 views

Self hosted FreePBX server, ZeroTier and Let's Encrypt?

I'm trying to follow this guide to setup a Let's Encrypt certificate for my FreePBX install. This is because to use Zulu, the mobile iOS client, you need a proper SSL cert. The thing is, my FreePBX ...
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23 views

Asterisk auto removed ps_contacts data when unregistered device

I tried changing the database from MySQL to SQLite3. Then I tried to use Zoiper to register using the registered pjsip number. Problems arise when unregistered devices. Data on ps_contacts is not ...
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17 views

Freeswitch + Asterisk servers over VPN

I have two offices which are connected via site-to-site VPN with static routing, VPN works great without any issues. Calls from Asterisk to Freeswitch works great, but calls from Freeswitch to ...
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30 views

Asterisk, display queue name or called number on the phone

Our Asterisk server will soon receive SIP calls on multiple numbers, so we need to know which number has been dialed to correctly answer on the phone : "Hello, this is Company-A" or "Hello, this is ...
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78 views

Connecting an old landline phone to FreePBX/VoIP?

I'm looking to hookup a landline phone with an RJ11 connector to my FreePBX server so I can make and receive calls using that phone. If this is possible, what type of adapter would I need?
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38 views

How is channel terminated if Hangup is not specified?

Consider the following example in asterisk/extensions.conf: exten => 1,1,AGI(my-script) exten => 2,1,Answer() same => n,Playback(tt-weasels) same => n,Hangup() Is it necessary to specify ...
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23 views

Asterisk Siptrunk Registration with multiple numbers assigned

I made a registration on my Asterisk in the sip.conf as following: register => myusername:mypassword:myusername@sip.test.com/720202020 So incoming calls works for 720202020. My question is can I ...
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151 views

DAHDI 3.0.0 not assign spans and cannot generate configuration

I'm setting up a new fresh installation of Debian 10, Asterisk 16 and DAHDI 3.0.0. I have specific the Digium TDM410P 4 port analog card (PSTN) and trying to configure it with asterisk and dahdi. ...
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71 views

How can I continue execution after a Dial from Asterisk

I want to continue execution after a dial (dial(SIP/name)) from the server Asterisk with, for example, a function playtones. How can I do that ? I did it with between two softphones but with directly ...
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57 views

How to use asterisk in a shell program

I want to use Asterisk in a shell Linux program more or less like this : asterisk -r && console dial 101 How can I do that ? Thank you.
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42 views

opensips + asterisk integration issue

I am trying to conigured opensips as registar and asterisk for media services using this doc: https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration so far everything went well ...
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112 views

Cannot add new System Recording: Unable to connect to remote asterisk

I having an issue when adding a new system recording .wav file (Admin > System Recordings). When I click on the Submit button. I got this dialog: There was an error, See the console for more details ...
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25 views

How to configure fop2 with xivo/wazo

I have installed wazo on Debian 9 and I also installed fop2 by following this documentation. When I try to connect to the connection manager using /usr/local/fop2/fop2_server --test, the the test ...
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143 views

Asterisk skips first DTMF

I configured an asterisk server to receive calls from one sip trunk and then dial out through another (my VoIP provider). Both trunks are configured with dtmf mode SIP INFO. The thing is: When I ...
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21 views

Asterisk: Get original DIAL Options in local extension

I have this Asterisk extension config which I am using to prepend a 0 to all incoming calls: [prependzero] exten => h,1,NoOp() exten => h,2,Hangup() exten => h,3,Hangup() exten => h,4,...
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23 views

how to configure xivo for WebRTC

I used this document to test the implementation of the webrtc on my website https://flashphoner.com/call-from-browser-to-mobile/ But I do not want to use their WCS5 server but my xivo pabx server. ...
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243 views

Firewalld rich rule doesn't immediately block IP address

I have Asterisk installed on a CentOS 7 server that uses the Firewalld firewall. To prevent brute-force attacks, I am setting up Fail2ban to ban IP addresses that enter incorrect endpoint credentials. ...
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25 views

How do I add static video to a call?

Problem: my conferencing software's SIP bridge only supports calls with audio and video. Now we can all agree that this is broken – in fact I already emailed my opinion about this to their support – ...
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245 views

One way audio with Asterisk 15 and a WebRTC client

I have two ways audio when calling from a WebRTC client to a mobile phone (through Asterisk), but when I call from the mobile phone to the WebRTC client the client rings, the call is established but ...
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49 views

programmatically initiating a call and connecting it to a conference on asterisk

I'm looking for a way to "script" a sequence where the following things happen in order Initiate a call to a predefined number When the call is answered, connect the call to a conference room. Call ...
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1answer
406 views

Asterisk nat=force_rport,comedia with routing

I have a setup where the phones are in 2 subnetwork, but only routing used between them not real NAT. 10.4.0.40 < Asterisk PBX 10.4.0.9 < Phone 1 10.1.0.66 < Phone 2 Phone 1 and the PBX ...
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712 views

Asterisk ignoring incoming calls with caller id

I have an Asterisk system which has been functioning perfectly well for about six months and now we want to add incoming caller id. We got the service from our phone company; we see the caller ID ...
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515 views

Asterisk 13.21.cert3 error loading res_rtp_asterisk.so module

We are upgrading (or trying) our asterisk servers from version 13.13.cert9 to version 13.21.cert3 and we are facing some issues. This are the steps followed to build it : Download and untar the ...
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35 views

Connecting a Fujitsu SS-170A phone to Asterisk

I have recently acquired a Fujitsu SS-170A VoIP phone in order to connect it to my Asterisk server. First things first, I am able to perfectly use softphones in the same network I am setting up the ...
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253 views

in process_sdp: Ignoring video stream offer because port number is zero (ONE WAY)

I have found dead ends for this problem on different forums and sites, it would be time to solve it for everyone's good. Given 2 devices: A cisco 894x IP phone with h264 support B softphone software ...
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204 views

Using Asterisk as a gateway to Provider

I have 2 sip servers on different LANs. Freeswitch and another is Asterisk. Asterisk sits on a VPN with a provider and who has provided DIDs. All users register on Freeswitch. How can I route calls ...
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118 views

Puppet Firewall string rate limiting not working for Asterisk

I have an Ubuntu 16.04 server and using Puppet 5 with Asterisk 14.6.0 running. I'm trying to implement the string rate limiting rules as described in the link below but it's not working. The CLI ...
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1answer
423 views

Asterisk PJSIP Jitter Buffer

I'm currently using FreePBX which has GUI settings to set Jitter Buffer for SIP, but not PJSIP. We have around 90 remote extensions using PJSIP and i would like to enable the Jitter Buffer for all as ...
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76 views

Server as phone behind eMTA

I've been tasked to research automation options for telephony modem/gateway device testing with a group of DOCSIS voice engineers. For each new DOCSIS device with an eMTA that we receive we need to ...
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575 views

asterisk playback sound when someone answers

This is an asterisk CLI question. I want to be able to play a sound byte with the asterisk "originate" command, but only when someone answers, like: channel originate *insertoutboundlinehere* ...
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55 views

Why asterisk listens on port 50360?

When I run: netstat -tulpn | grep asterisk I see the following: tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN 12337/asterisk tcp 0 0 0.0.0.0:5060 ...
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301 views

Asterisk JITTERBUFFER function details

Asterisk v15 documentation states that JITTERBUFFER function is used to "add a Jitterbuffer to the Read side of the channel". Take this dialplan example: [from-pstn] exten => 1234,1,Set(...
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215 views

Asterisk: Configure the Server header field

How do I modify the Server header field in responses by Asterisk? By default, it includes the release number and the package name: Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3 Server: Asterisk PBX 11....
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725 views

Disable IAX module on asterisk. Only allow SIP

I wanted to see on what ports my ubuntu machine was listening on; therefore, I ran the command netstat -tulpn. When I ran that command I found out that I was listening on port 4569. That is being ...
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996 views

When secure calling with asterisk (SRTP), why are client certificates needed for SIP devices?

I just added security to Asterisk by following this tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial#SecureCallingTutorial-Keys Note that asterisk does not install by ...
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438 views

Asterisk uses the wrong variables from sip.conf

I completed this tutorial in order to make secure calls with asterisk. Secure Calling Tutorial | Asterisk Project Wiki I am running asterisk version 13.19.2 on Ubuntu version 16 (debian) and as soon ...
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959 views

How to enable asterisk Call Completion on Busy Subscriber (CCBS)?

I cannot enable Call Completion on Busy Subscriber (CCBS) on asterisk, witch is part of the Call Completion Supplementary Services (CCSS), as is the Call Completion on No Response (CCNR) feature. ...
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356 views

rSyslog - Disable to push custom logs to messages

I sending my Asterisk's logs to Graylog. (its working fine) But the rSyslog write every asterisk's log to /var/log/messages not to the /var/log/asterisk/full. Asterisk logger.conf: [general] ;...
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413 views

Asterisk - Outgoing Caller Name on Client

When placing outgoing external call using SIP client Bria. The phone displays our caller ID as the name of our calling party. Looking over the log, it shows that it came from this line -- Executing [...
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829 views

Causes of RTP jitter at the server

Investigating some call quality issues (0.5 – 1 second dead spots in calls) I took a packet capture of a phone call between two extensions on the same PBX. Since I was capturing from the PBX, I was ...
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2k views

Asterisk compilation issue

I've downloaded source code of Asterisk from http://downloads.asterisk.org/pub/telephony/asterisk/ I'm getting error while compiling this from source code in Ubuntu 16.04.1. Please suggest ...
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409 views

“register” to asterisk with any password

I have set up an asterisk server on a non-important VPS to test something; there is no sip trunk, so it can't call anywhere. I see a lot of attempts to register; I'd like to allow them to register to ...
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444 views

FreePBX - How to re-write forwarded “follow-me” caller ID?

Sam calls Julie's direct dial line and because she's not at her desk (and because of a follow me rule) the call is forwarded to her mobile. FreePBX forwards the call through the correct outbound ...
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164 views

Asterisk (VOIP) port forwarding on TP-LINK [closed]

I have local asterisk server on UBUNTU 16.04. It's address 192.168.0.101:5060 and I would like to be allowed to talk with other users when I am outside local network. I tried THIS but it doesn't work. ...
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469 views

“channel allocation failed: refusing due to active shutdown” warning

Our phones are down with the warning above when I try to run command show channels. Is a core restart was done, how long before phones back up? Or is there something else that needs to be done to ...
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414 views

Asterisk / FreePBX: Calls to internal extensions require users to press Dial

Using FreePBX 14 and the Cisco SPA504g phone, when my users dial outbound US number (e.g., 15555551212) the calls automatically dial. However, when the dial 555 for an internal extension, they have to ...
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596 views

Asterisk Confbridge conference. How to avoid eavesdroppers?

Trivia: I'm trying to implement a conference call management functionality on my Elastix server. The original problem was in the Elastix's unability to invite all participants in the conference via ...
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1answer
820 views

CentOS 7 Asterisk + FreePBX Install:

I attempted to run a ./install -n on my FreePBX install and got: ===== Checking if Asterisk is running and we can talk to it as the 'asterisk' user...Error! Could not determine Asterisk version (got: ...
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2answers
229 views

puppet iptables string rate limiting

Does puppet firewall support the string module? I need help to implement the following iptables rules in puppet but can't find any documentation or examples on how to do this. iptables -I INPUT -p ...
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265 views

New Asterisk 13 & 14 missing Application “Read()”

I just upgraded from Asterisk 11 (Elastix) where this script was working to Asterisk 14. I am getting a slew of errors that are surprising. It appears problem is that there are some "default" ...