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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18 views

Working on video configuration suitable for webrtc and sip/udp user agents. One way audio and rtcp timeouts

WebRTC-WebRTC call works well, but SIP-SIP and WebRTC-SIP no: [Oct 21 18:27:22] DEBUG[59]: pjproject: <?>: icess0x7f66780b0c98 Controlled agent timed-out in waiting for the controlling agent to ...
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23 views

Status puplication on asterisk

Is it posible to make asterisk publish user agent statuses like online, dnd, etc..? Then i change status on Linphone client i get folowing warning: [Oct 22 12:30:47] WARNING[46]: res_pjsip_pubsub.c:...
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71 views

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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133 views

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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98 views

Setting up Asterisk + webrtc

This question has already been asked here. However based on a comment there, I am posting it here. I am trying to set up Asterisk to work with webrtc.. On the client side I am using sipML5. This is ...
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11 views

How to retreive IAXVAR() variables in YATE?

I have an Asterisk 18 PBX as IAX2 (trunk) server in a VPS. In my office I have a TP-Link (OpenWrt) router installed Yate (Yet another telephony engine). I cannot install asterisk because it does not ...
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26 views

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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24 views

Asterisk - Conferencing MulticastRTP and SIP channel together AFTER commands on run on the SIP channel

I have a PA system in the building that requires dialing an extension and then #00 to signify "All Call". Also, the phones can do pages through multicast. I'd like to tie these two systems ...
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52 views

PJSIP Call file

I use call files to send calls on asterisk. with the change to PJSIP the old script no longer works, can anyone help with a functional call file that works with PJSIP? my call file channel: PJSIP/...
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41 views

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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35 views

Asterisk AGI -> Bash-Script -> causes choppy audio

for the last days i replaced 90% of my extensions.conf to an external AGI-Bash-Scripts. After that, the audio was choppy in both ways for all running channels, when the AGI-Script gets executed. After ...
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129 views

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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112 views

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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87 views

Asterisk kicks me out of the cli

i'm struggling with an issue in asterisk: it suddenly kicks me from the CLI and interrupts any call when that happens. Also the followng messages are displayed: Asterisk cleanly ending (0). Executing ...
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38 views

asterisk AGI PHP Ignored

I have written exten= 6001,1,AGI(pk.php) content is : (and i tried other samples) #!/usr/bin/php -q <php? set_time_limit(30); require('include/phpagi.php'); error_reporting(E_ALL); $agi = ...
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57 views

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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28 views

Forwarding Caller ID info to a specific IP address

I'm having an issue where I'm supposed to log all the incoming Caller IDs to a specified IP address in the Elastix environment. Can you suggest anything workable as a solution or to what should I ...
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46 views

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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62 views

In Asterisk, how can I screen calls inside of a queue?

When a call comes in, it is routed to a queue, and that queue goes to a few places: Desk phones Cell phones Call centers The cellphones and call centers may "answer" the call without ...
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76 views

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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159 views

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
177 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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340 views

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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2answers
573 views

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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286 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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116 views

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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616 views

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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729 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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57 views

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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1answer
810 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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57 views

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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1answer
494 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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275 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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295 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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114 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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942 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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352 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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157 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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1k 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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55 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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1answer
180 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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59 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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1answer
1k 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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1answer
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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1answer
1k 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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67 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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624 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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986 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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163 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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877 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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