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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Authentication error in trunking between two asterisk

i'm creating a connection between two asterisk servers. I want only make calls from right side server to left side server, but i receive an authentication error. I can´t find the problem. The Error ...
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Asteris No compatible codecs, not accepting this offer!

I have installed an asterisk it repeats this statement, again and again, my server is overloaded because of these errors. [Sep 19 18:34:23] NOTICE[1641][C-00000398]: chan_sip.c:10995 process_sdp: No ...
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connecting two SIP phones from different LANs over ZeroTier openwrt

i make site to site conncetion by tp-link n600 openwrt zerotier and no problem , i can reach every thing in two sites and ping ok but i have in site 1 Issable and in site 2 sip phone ------------------...
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Problems setting up a secure WSS connection for Sipml5

I have problems with setting up a webrts connection with sipml5 through an asterisk. When I check the status of https through the asterisk console, I get a response that it is up and running. But when ...
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My asterisk server can't read extensions.conf

I've installed asterisk on ubuntu 22 on virtualbox everything is ok when voip call comes to my server it says me : rejected because extension not found in context 'default'. my sip.conf file : [1060] ...
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Get Asterisk IVR inputs to Linphone user

Let's say someone creates an Interactive Voice Response (IVR) system using Asterisk, that prompts the caller to input his customer number using his dial pad, e.g. 1234. How does one get this ...
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Fail2ban asterisk directly ban

I have the following fail2ban asterisk settings set. [asterisk] port = 5060,5061 action_ = %(default/action_)s[name=%(__name__)s-tcp, protocol="tcp"] %(default/action_)s[...
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Asterisk shows No matching endpoint found

I have planned to implement Asterisk SIP server for testing eMTA calls. I don't have eMTAs so I decided to start with Linux Soft client and then, when I will have an eMTA and physical access to the ...
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Migrating Asterisk VoIP to new network

I've an Asterisk server with 4 IP phones connected to it, and gotta move them to a new network (IP range). When changing the IPs and gateway of the server and the IP phones, the phones stop working ...
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fail2ban not creating jails in CentOS

I'm trying to create a jail in Fail2Ban to protect a Asterisk PBX. I already did as suggested and made a copy of jail.conf named jail.local. In jail.local, I set up a asterisk jail as follows: [...
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How to enable queue_log on the embedded Asterisk of a Yeastar MyPBX?

I have an old PBX from Yeastar that contains an instance of Asterisk 1.6.2.6. How can I enable the queue_log setting and have it persist across reboots?
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Asterisk: pri to pri connection. Green light but a lot of hdlc errors

I want to connect(for learning, nothing serious), two asterisk server via ISDN PRI. The first server will act as "telco" so is pri_net, the second server is the cpe, so I will use pri_cpe. ...
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VOIP, How to setup on Vicidial?

I am runing a vicidial system on a computer , so far so good. I have done basic configure like creating campagnes, users, users group, phones. And if i make a call inside the network it works. But now ...
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asterisk - what's difference between format_* and codec_*?

When building Asterisk, there is two different entities related to audio formats - formats and codecs. What's difference? For example, I need a support only for alaw+ulaw+g.729 on voice traffic itself,...
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No RTP engine was found. Do you have one loaded? Asterisk-18.10.1

I have been trying to install Asterisk-18.10.1 version on my ubuntu(20.04.4) running inside VM. I was able to maintain connection from GoTrunk SIP endpoint and Zoiper as softphone. Followed https://...
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How to change pickupexten on asterisk 11

I'm looking for a way to change ‍pickupexten in Asterisk 11. Is there a way? The features.conf: pickupexten = *8 pickupgroup=1 pickupsound = beep pickupfailsound = beeperr I want to change *8 with ...
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How to create/verify client certificate on Asterisk? any way to verify "common name" field?

Current configuration: We have a PBX (Asterisk) on external IP to allow all SIP clients for registration/calling etc. Currently, only port 5061/tls is opened and client certificate is required. Pjsip....
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Forwarding SIP headers with asterisk (PJSIP)

I'm trying to forward a specific incoming header to the other leg of the call, but can't figure out how to pass the value of the header in the incoming leg to the pre-dial handler [addheaders] exten =&...
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command that cannot work in background (asterisk)

am trying to run script in background which contain command such like "asterisk -rvd >> xyz.log", when I run it directly it work well but when run it with any background way (service, ...
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asterisk not recognizing answer from sip trunk

I have an Asterisk server (15.5, FreePBX) with three SIP trunks from different providers configured, two of them are working fine while the third for every call keep sendind the invite despite the ...
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Asterisk automatic hang up when I try to call

I just set up a server with Asterisk set up but when I try to make a VoIP call the line automatically drops... I use CommPeak as a SIP provider and Bria for the interface Here's my sip.conf: [general] ...
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VoIP one-way audio, only when call initiated from one side

I am setting up some new switches and VLANs and I am getting trouble with our pre-existing Asterisk VoIP set-up. Most calls work ok. Some get just one-way audio. I tried to narrow it down to this ...
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Asterisk/FreePBX queues no longer working

Incoming calls go straight through to failover regardless what other settings are. I have deleted the queue and rebuilt it, nothing seams to fix it... Asterisk 16 Freepbx 15 Debian 10 There is a ...
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adding modules to installed asterisk - alsa.so not found

I have set load= chan_alsa.so and i got this error ERROR[77064] loader.c: Error loading module 'alsa.so': /usr/lib/asterisk/modules/alsa.so: cannot open shared object file: No such file or directory ...
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Asterisk - Position Announcement to First Caller

When Asterisk announces the position of a caller in the queue, it plays queue-thereare, followed by the caller's position, followed by queue-callswaiting. We have custom messages for each of these ...
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tel: URI support in Freeswitch(for outgoing calls)

I have integrated a Freeswitch instance with an external SIP server and calls are working without any issues. But now I need to change the SIP server which only supports tel: URI. In the case of ...
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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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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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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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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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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: 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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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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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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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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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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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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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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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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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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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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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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