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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Asterisk - detect DTMF during call, emit a different DTMF code

I inherited an old Asterisk system v. 1.4.39.2. Its dialplan is created by a GUI called IpBrick, (vaguely similar to FreePBX), but I can tweak most things through includes. There is a door phone that ...
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Asterisk dialplan pattern generator

I was wondering if anyone knows about a shell script or Pyhton script, or whatever language, some kind of command line script that given a set of phone numbers, the script will then generate the ...
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How can I build my own SIP trunking with Asterisk?

I want to create SIP trunking. I am using Asterisk 16.13 on Ubuntu 20.04. I'm a student that didn't have experience with PBX and SIP trunking. I would like to know if someone knows how to make SIP ...
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I get 404 on all asterisk pages

I want to use asterisk gui, using the internal webserver in asterisk. But I get only 404 not found. http.conf: [general] servername=Asterisk enabled=yes bindaddr=0.0.0.0 bindport=8088 ...
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FreePBX dropping calls from Twilio for unknown reason

I have a single FreePBX system connected to Twilio's Elastic Trunk service. Local server IP is 10.0.3.1. Sometimes, when a call is placed, the call comes in, goes through our system, and routes ...
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Audio issue in public network with Asterisk 17

I have just installed and configured Asterisk 17 in a desktop PC running Ubuntu 18.4 My Asterisk and one of the clients using Zoiper Softphone are behind NAT. Another Client is an iPhone running on 4G ...
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1answer
26 views

Asterisk: “in call” queue member status, while actually no calls

My asterisk won't direct inbound calls to certain queue members, because it thinks that they are busy receiving another call. But they are not. They are waiting for a call (register and waiting), or ...
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33 views

Asterisk ARI: Get the application name of a channel

I try to access my installation of asterisk via ARI. When I use curl on the endpoint /ari/channels (GET), a list of all existing channels and some detailed information are being returned. According to ...
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(Asterisk)How to troubleshoot incoming voicemail calls with AMD

guys, I would like to know if there is some way I can insert a log (NoOp) in the context of the asterisk where I can inform which values were detected by AMD for each connection. Example: Call X was ...
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122 views

Can't register asterisk to sip provider using PJSip module

I have recently set up asterisk server on the Azure cloud, able to connect to sip provider (using sip module), and able to place a call successfully, thanks to all the support I found here/google/(...
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Asterisk dail tone while waiting for paging partner

I am using the asterisk audio socket extension (https://github.com/CyCoreSystems/audiosocket) to get a live stream of an onging phone call to another device (simple tcp socket). Let's say there are ...
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Asterisk, Loadbalancing and SSO

I am trying inverstigate/evaluate asterisk with reposct to loadbalancing and SSO. We have in house solution for SSO adhereing to standards , bulit with spring Framework, can we use SSo with asterisk ...
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66 views

Low audio volume on sip calls

Recently a new business moved into our premises and we added them to our FreePBX switchboard and ported their numbers to our VOIP provider. All the routing IVRs and everything has been set up. They ...
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asterisk -vvvvvrx 'core show channels verbose'

The output of the command asterisk -vvvvvrx 'core show channels verbose' shows total number of calls processed. That number is an indication of the calls processed since the service is running or last ...
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Issabel-Asterisk- How to handle multi-digit inputs on IVR

I have an Issable server running and integrated with Cisco Call Manager correctly. I am using Issabel server as my IVR system to answer calls without any direct extension callings. My Scenario is as ...
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80 views

how to run multiple instances of asterisk on one server

hope everybody is safe during challenging time i have a task to automate our demo installation process. Our software is mainly a GUI that operates with asterisk via AGI and stores all data to MySQL / ...
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Large Jitter Spikes on a Small Percentage of VoIP calls

On about 5% of our customer's calls, we see large jitter spikes and high delta #s that have caused an noticeable audible impact on call quality. (Stuttering/Breakups/Robotic Audio). We know this from ...
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Where Is Asterisk FreePBX Video Call Recording File

I would like to know where can I find (if it does exist) the video file for a call with video. I already have all the things running properly: Call recording enabled for extensions as Force Call ...
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FreePBX / Asterisk: use inbound routes to block spammers/hackers

My FreePBX / Asterisk configuration was recently forced into allowing both anonymous inbound calls and SIP guests. So of course we're now getting blasted with spam/hack attempts. They show up in the ...
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Asterisk Integration Issue with CRM

I have configured V-tiger CRM. I want to integrate v-tiger CRM from Vici-dial asterisk but in v-tiger CRM there are not showing phone calls extension in extension store. I have configured the call ...
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346 views

Asterisk: Using Queue() in the h extension

I'm trying to get users to be able to record a message, hangup and have the call continue and dial Queues and playback the recording. I've gotten most of the way there, but right now when I call ...
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Initial one way audio with SIP trunk

I have an Asterisk server sitting on my network behind a pfSense firewall, it has two trunks, one for my household provided by my ISP using PJSIP and the other for my business provided by a third ...
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Wildcarded Asterisk SIP Resources

I am creating a call queue of sorts whereby an incoming PSTN call rings multiple SIP extensions. That part is easy using Dial(SIP/resource1&SIP/resource2&...&SIP/resourcen). The issue is ...
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372 views

Asterisk WebRTC outgoing call delay

I run an Asterisk 16 installation and a WebPhone based on SIP.js. Unfortunately, I often don't hear the first few seconds when I call someone. But everything is fine with incoming calls. The Asterisk ...
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743 views

freepbx/asterisk “All circuits are busy” new install

This has been an issue with every install I have ever put together and could google the answer quickly. However, I haven't built a freepbx for over 2 years and cant recall how to resolve it, and can ...
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Can't connect PC to an Asterisk Server (IP)

I have to connect an external network to an Asterisk Server who is available with an external IP. I tried to ping the server from the network but it's not working. Meanwhile, it's working for every ...
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198 views

Why my Fail2ban regex doesn't find matches?

I have tried many regular expressions in fail2ban config, but it never return any matches. Line example: [2019-12-10 10:45:38] NOTICE[15077] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '<...
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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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146 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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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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620 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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163 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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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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591 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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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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153 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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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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976 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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1k 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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918 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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51 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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475 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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528 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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140 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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618 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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89 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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837 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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71 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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454 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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330 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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