Questions tagged [asterisk]

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

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Router won't forward ranges of ports to Linux server. Is there a software solution?

I need to forward a large range of ports to a Linux machine running Asterisk. The machine is behind a household router. The router can be successful configured to forward single ports to Asterisk, e.g....
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1answer
19 views

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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15 views

Can you use a regular SIP line as a trunk

I have an asterisk box that can route calls internally. I am trying to determine if I can connect the asterisk box to standard SIP line like AT&T's Digital Voice. I understand that I would only ...
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21 views

Setting up a computer as a phone station via my router's telephone ports [closed]

First off, I should note that this is my first time working with telephone hard-/software beyond simply using a phone, plugging it in or setting up a router. In addition, I'm aware I could just use ...
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74 views

How to make asterisk call file as user account or extension

I am trying to figure out why the callerid works when I call with twinkle through asterisk but not when making a call file. The callerid with the call file is always unknown. Twinkle account is setup ...
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1answer
45 views

Asterisk 13 & Pocketsphinx Segmentation Fault

I've built a voice recognition system for german digits using asterisk 13, pocketsphinx and a plugin and server (based on astsphinx) to connect asterisk with pocketsphinx. When testing the system with ...
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1answer
50 views

Asterisk configuration, need to use a different internip for each trunk

I'm currently configuring my asterisk server, that has two trunks, one for the incoming calls, one for the outgoing calls. For each trunk, I would need to define a specific internip. So far, the IP ...
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33 views

Asterisk Call Data Record Logging via Adaptive ODBC to IBM DB/2 not working

I'm running a stock Debian 9 with the accompanying Asterisk packages (Version 13.14.1). To save loading time and memory consumption, I don't autoload modules in modules.conf but edit the list of ...
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1answer
98 views

Asterisk: check original Caller ID on transfer

I am using Asterisk 16.5.0 with usecallmanager patch (which allow me to use Cisco phones features). We have a number of ConfBridge rooms, which are supposed to be called by employees only. Some ...
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1answer
34 views

How to omit a bad PSTN line in Asterisk

I have an Asterisk/Dahdi server that serves as a PSTN software PBX running on Debian with a Digium 1TE134F card. The setup has been up and running for sometime but we noticed that one of our lines is ...
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40 views

Kubernetes behaves differently than similar setup in docker, what are the differences between the networks?

I have a docker image which is producing audio (SOUND image in the attached drawing) and an image containing and asterisk server (* image). When I host these on my local server and call them with my ...
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1answer
106 views

Asterisk, IAXModem & HylaFax: No carrier detected

I try to get a HylaFax + IAXModem + Asterisk Setup on a Debian 10 server running. The server is directly connected to the Internet, the firewall is disabled (no rules set up). The only thing that ...
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1answer
243 views

How to configure instant messaging in freepbx 14 & Asterisk 13

i have a trouble to configure instant messaging using freepbx 14 and asterisk 13 , i want that two sip clients can send and receive messages using their soft phones on smartphone and desktop , can ...
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36 views

No periodic announce in realtime queue with Local member

I have problem with queues in Asterisk 11 [queue1] (static in queues.conf) periodic-announce-frequency = 15 periodic-announce = queue-periodic-announce member=SIP/101,1 member=SIP/102,1 member=...
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0answers
21 views

SIP trunk configuration on FreePBX

How we can configure SIP trunk on Asterisk and FreePBX to re-route the incoming call from mobile/landline over internet.
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33 views

Query Regarding routing a normal phone call through PC

I have an urgent requirement of routing a normal phone call to PC. There are additional requirements such as – Call will come from landline/mobile number be to a fixed number, and this call should be ...
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1answer
81 views

Sip Trunk Peer issue with 2 asterisk servers

I am getting the unauthorized error while trying to place a call: [Sep 3 16:20:19] Asterisk 13.27.0-vici, Copyright (C) 1999 - 2014, Digium, Inc. and others. [Sep 3 16:20:19] Created by Mark ...
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1answer
45 views

Random hop in IVR

I have an Asterisk 13.23.1 server with freePBX 10.13.66-1. My problem is this, in one IVR when the caller puts a no valid destination (enter a number not configured in the IVR plan) it should go to ...
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39 views

Linksys SPA921/942 registering with the wrong extern IP on an Asterisk

I have a Linksys SPA921 and a SPA942 behind a NAT that register with the wrong IP on my asterisk which is running through docker-compose on a Google Cloud instance. I had to reboot the instance a few ...
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67 views

Asterisk map DID to extension in a dialplan

I run FreePBX 14 (Asterisk 16.3.0) and would like to make use of the SMS feature. They provide a basic setup at https://wiki.voip.ms/article/SIP/SMS_with_FreePBX However, this set of instructions ...
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0answers
694 views

Asterisk: chan_sip.c:4321 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data

I have a setup of Asterisk PBX running on a Raspberry Pi 1 (IP 192.168.2.2) and it runs a few local IP phones just fine. I recently purchased a Grandstream HT813 gateway (IP 192.168.2.3) to connect ...
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1answer
164 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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2answers
309 views

Asterisk 16 LTS & PJSIP; hello world works but no sound coming from endpoints

I have recently set up an Asterisk server with version 16.4.0 LTS. I am completely new to asterisk yet I have managed to set up the server with the service and it runs smoothly among LAN users and it ...
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1answer
62 views

Supervisor don't see playback recording in freePBX UCP dashboard

My IT Head surely change the default location where call recordings are being saved, as a result Team Leads can no longer view playback button in UCP, they cannot do call listening. I believe that ...
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1answer
1k views

Asterisk 16.4 pjsip trunk registration

I'm not an asterisk expert, and I'm stuck at this moment. I'm trying to setup an asterisk box with realtime. Most work, and my endpoints are able to make calls between each other. But I need to setup ...
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163 views

asterisk ODBC Data source name not found debian 9

I've been trying to get odbc working with asterisk. As of now, I think the fault is with ODBC/driver/packages as I am not able to even run isql command and get output: Error : root@odbctest:/home/...
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0answers
95 views

Asterisk jitter buffer on receive side of B-Leg (device side)

I've been trying to apply a jitter buffer to the B-Leg or device side of calls for some time now without any success. I've set the jitter buffer to something very large just to test, using an echo ...
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1answer
111 views

Asterisk how to call by names instead of numbers

I haven't found anywhere clear sample how can I setup alphanumeric alias for internal numbers. For example I am calling now to 504@ip. How can I make it possible to reach 504 number by john@ip by ...
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1answer
662 views

Asterisk RTP re-invite not working

I’m experiencing the following scenario in Asterisk, and was wondering if you could help: Asterisk source IP sends call to destination signalling IP Destination signalling IP answers call, specifying ...
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2answers
50 views

Add “+” to Cisco SPA122 (latest firmware)

I use Twilio, and I'm required to use the full ISO spec "+" before all phone calls. MY PBX is Asterisk 13.26.0, connected through private network. I have a Cisco SPA122, with the latest firmware. ...
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1answer
272 views

Hot to force Asterisk to write date in UTF-8 charset?

CentOS Linux release 7.6.1810 (Core) Kernel: localhost.localdomain 5.0.7-1.el7.elrepo.x86_64 MariaDB: Ver 15.1 Distrib 10.4.4-MariaDB, for Linux (x86_64) using readline 5.1 Asterisk: 16.3.0 built ...
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1answer
23 views

Cron job schedule to strem via RTP audio file

I have Asterisk 1.8.28 based PBX and I want to Intercom or Page some audio files at different times daily, I have Cisco SPA508G working as RTP receiver connected to my amplifier and bunch of speakers, ...
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1answer
1k views

can't run asterisk, no /etc/asterisk/asterisk.conf

i'm trying to install asterisk on my server, I follow to this sample But, in the end of installation, when I call sudo asterisk -vvvr it shows me Unable to connect to remote asterisk (does /var/...
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1answer
17 views

Is it possible to change files with recordings (answers) in freePBX, Asterisk during talk?

In IVR (freePBX) there are answers recorded that are played when someone is calling , but is it possible to change those answers via REST API or switch recordings "on fly"? I am looking for a way to ...
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2answers
329 views

Is it possible to decrease the transmit/receive volume in Asterisk below normal?

I am playing around with changing the volume in Asterisk. There are a few applications where I would benefit from this. Most of them are to increase the volume but there are also some scenarios in ...
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1answer
2k views

asterisk: error while loading shared libraries: libasteriskssl.so.1

I am using Ubuntu 14.04 and followed the instructions to install asterisk from this link Ubuntu 14.04 Asterisk server installation All worked well until I run asterisk and get a shared lib error. ...
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1answer
231 views

Asterisk sip.conf in MySQL database

I'm able to include a phone inside a database for realtime usage. So, this code (from /etc/asterisk/sip.conf): [phone] type=friend username=phone secret=12345 host=dynamic disallow=all allow=g729 ...
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1answer
975 views

How to setup Asterisk NAT on docker for Windows

I need to setup a docker container running an Asterisk PBX. The problem is that I receive the wrong ip in sdp replies. In fact Asterisk is using the container internal IP. The second problem is that ...
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0answers
369 views

Asterisk not adding CDR record in DB for call to unregistered peer

I configured Asterisk CDR backend to PostgreSQL, and everything is working fine for calls between two registered peers. The problem is when a user calls an unregistered peer. I want Asterisk to ...
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0answers
127 views

On a PBX (Asterisk) determine which phones are connected

I am using Asterisk as my PBX. On my sip.conf I used to have this option enabled: [general] ... qualify=yes more info about qualify can be found in HERE. Thanks to that option I was able to tell if ...
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2answers
461 views

Converting a 2 channel stereo g729 file to individual single channel pcm encoded file

Whenever I try to convert a G729 encoded two channel file into mono channel pcm encoded wav files I get the following error Input #0, wav, from 'g729stereo.wav': Duration: 00:00:07.07, bitrate: 16 ...
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2answers
2k views

Asterisk NOT FOUND answer on OPTION request

Remote server send me OPTION package, but my asterisk server send "404 NOT FOUND" response. Is there an option in sip.conf or something else? sip.conf [remote-server] type=friend host=10....
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1answer
2k views

Asterisk Trying to play back audio file “File /path/to/file.wav does not exist in any format”

i'm trying to rig an application to send out premade calls. I'm trying to get asterisk to play back wav/mp3/gsm files (I compiled it with the proper extensions and they're loaded) but when ever I ...
3
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0answers
132 views

Setting CallerID on outgoing ISDN calls? [closed]

Does anybody have any experience of settings the CallerID on outgoing ISDN calls? We are trying to set the caller id to one of the available DDI's on our circuit but without success. They worked on ...
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0answers
52 views

Understanding Yealink VOIP phone configuration

I know how to connect my VOIP phone to asterisk. My asterisk server sometimes goes down so I will like to understand better how can I use a failover server. This guide: http://download.support....
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1answer
342 views

How to add new CID Superfecta source in FreePBX?

I don't find information about how to add a new CID Superfecta source (my own custom programmed one) in FreePBX. Documentation doesn't say and there is no obvious method to do it in the FreePBX ...
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0answers
2k views

How do you fix “No Trust List Installed” error on a Cisco 7942G VoIP Phone?

I have a 7942G. I'm trying to connect it to Asterisk. Looking at Settings -> Status -> Status Messages, I see this: Error Updating Locale SEP############.cnf.xml No Trust List Installed I'm only ...
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1answer
131 views

How do I make music interactive using Asterisk?

I recently started my own computer repair business & use FreePBX/Asterisk for my phone system. Just wondering if it's possible to make my music on hold interactive. I would like to let callers ...
2
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1answer
380 views

How can two sip users talk to each other by asterisk?

I am using asterisk 13 on centos 6. Now I have add two users in pjsip.conf as below: [transport-udp] type=transport protocol=udp ;udp,tcp,tls,ws,wss bind=0.0.0.0 [6001] type=endpoint transport=...
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1answer
326 views

How to retrieve and save Asterisk phonebook entries via ARI?

How can I retrieve/update/add Asterisk phonebook entries via ARI? I'm looking at ARI documentation https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ARI, but I don't find anything about the ...