Questions tagged [pbx]

A private branch exchange (PBX) is a telephone exchange that serves a particular business or office, as opposed to one that a common carrier or telephone company operates for many businesses or for the general public.

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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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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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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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For SIP trunks, is it possible that DTMF talk-off could mute a call?

My company sells/supports Allworx SIP PBXs. We have a weird edge case with one of our customers who use PRI for dial-tone. On outbound calls, if a user happens to receive an SMS on their iPhone, and ...
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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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netsed transparent proxy and server on the same system

I want to use netsed to alter the incoming SIP traffic (UDP port 5060) on a PBX server which is running on a linux system (debian 10 stretch). In the first step I simply tried to set up the phone to ...
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850 views

3cx firewall error message when testing ports

i just deployed a 3CX pbx VM, once i try to run the firewall test i get this error message with all the ports that are tested: "testing port 9000... full cone test failed (How to resolve?)" this ...
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Capture the VOIP calls receive to a sip phone that send Notificantions to an API on a call

This is whats I'm trying to achieve I want a SIP phone or (is there is any other way I'm open too) to take the calls it receives from a VOIP provider take the information related to the call when ...
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481 views

IIS as a reverse proxy for a server with many ports

I have a Windows Server 2016 machine with sales & accountant software running on it. I want to deploy a telephony PBX server, running inside a linux Virtual Machine (via VMware). It's running on ...
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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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1 answer
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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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Add Linux VPS (cloud VPS) to Local Network using VPN or IPSec

I have a Linux (Debian 8) VPS (cloud based VPS, not on Google Cloud platform, but that should not matter) on running a software PBX system. I want this remote cloud based Linux VPS to appear on my ...
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277 views

HA on CentOS6.5 constantly switching

I have 2 server Asterisk on centOS in HA group with peacemaker.The cluster is in active passive mode with configured resource groups.The problem is from some time the server switch from the primary to ...
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Does this follows the SIP standard?

We are currently using a Swyx PBX which connect using SIP to a OneAccess box (which translate ISDN <-> SIP). We have trouble to make it works. Here's the deal: The Box uses the SIP protocol on ...
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plug in an asterisk based solution into my PBX network

Introduction my company has a 40+ phone system with both DECT and wired phones. the wired phones, coming from different periods have either UTP wiring or classic 4wire phone cabling. NO SIP/VoIP is ...
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Migration to Skype for Business

I have a new customer that is looking to user Skype for Business with their Office 365 subscription as their phone system. Is there a good resource out there that can help me guide them on what needs ...
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Use asterisk as client to act as a soft phone

I have an account with vonage and I am able to make phone calls from my computer entering the following information into my soft phone (zoiper): When entering that information (domain,usrename and ...
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HP 4120 phone and Asterisk

I got a task to set up a cheap PBX for HP 4120 phones. I want to try Asterisk, but I can't find any info on connecting HP 4120 to any SIP PBX. So here's the question: are HP 4120 compatible with ...
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Trouble moving freepbx and a2billing

I'm having trouble moving from FreePBX 2.9.0.15 and A2Billing 1.7.1 to FreePBX 2.11.0.43 and A2Billing 2.2.0 I've transfer the A2Billing data via the backup and restore feature without any problem. ...
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How to test a second PBX securely?

I would like to test a second PBX software without causing damage to current company PBX. For instance, i would like to be sure that if I install this second PBX, all my company phones don't start ...
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475 views

Why do we even need a PBX software like Asterisk?

I am starting out with VOIP and configuring softphones. When we sign up to a SIP provider like FlowRoute we get the credentials for a SIP Proxy. Can't we just use the SIP registration details in a ...
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Is someone is trying to access our IP-PBX

We run asterisk on a freePBX distro. The system is only up for one day. When I logged in to asterisk from the terminal the system was generating a lot of unexpected messages just like the following. ...
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Elastix 2.5 - Change settings via WEB GUI does NOT take effect

My PBX system is currently running Elastix 2.5, which has been installed & configured about three months ago. For maintaining purpose, I have used etckeeper to tracking changes in /etc folder, ...
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using panasonic ip phone kx-nt511abxb over vpn

we are using panasonic pbx system kx-ns500bx. now we started new remote site and planned to use two extensions there through vpn. already we have cisco asa 5505 ipsec vpn for using accounting ...
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High availability Asterisk options?

We currently run FreePBX on a single node. I'd like to have a cluster of asterisk instances for sharing the load, but mainly for failover. I'm curious how other people have solved this problem. ...
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Microsoft UC and Skype for Business

I new to the Hosted PBX and VoIP systems. Currently the company I work for has an old T1 phone system they are looking to move away from. We have Skype for Business and Office 365. Basically my ...
2 votes
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266 views

Asterisk Transfer call to next extension if previous INUSE

I am trying to transfer call to next extension if previous is using (INUSE) or call is in progress. I tried to use EXTENSION_STATE(extension[@context]) to find the status as follow: [sales] exten =&...
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What is the best way to divert an IP address between two remote locations?

I would like to ask the community thoughts on the following. I have two data centers, one up 'Norf, and one down South Both are isolated in terms of network, but both have public IPs on a dedicated ...
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556 views

Asterisk dialplan several apps simultaneously

I am designing an Asterisk 11 Dialplan for a call center. I am facing a problem in dialplan. In order to achieve the desired configuration , i think it is necessary to run 2 applications at the same ...
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UK config for SPA232D

I have recently acquired a SPA232D to integrate some VoIP to a system using asterisk. Clearly, it needs to be configured for UK (as distinct from US, AU, other European etc) telco's. There's are some ...
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Asterisk WaitForSilence NEVER detects silence

I am trying to use my dialplan to play recordings with WaitForSilence to make sure it wait until the person is done speaking or the message is left on voicemail. However, it doesn't seem to wait for 5 ...
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One-way audio problems with SIP over AT&T

I have a customer who has an VOIP PBX connected to a Level 3 fiber connection. He has offices all across the country using different ISPs. Two of those offices use AT&T, both in different states. ...
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Cisco IP Phone TFTP Error/Timeout

I'm trying to get Cisco 7942G IP phone working on FreeBPX setup. It has issues with getting things over tftp. I tried hard reset thing and I'm getting this in tftpd log: [root@localhost ~]# tail -f /...
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Voicemails cutting off in freeswitch

We are using a freeswitch PBX. Any incoming voicemail getting left is being cut off after 60 seconds - i.e. the call is actually getting disconnected at that point. Regular calls are not, just ...
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Asterisk IVR setup: disallow direct dial for some extensions?

We've got PBX to configure. It's based on Elastix 2.4.0, freePBX 2.8.1 and Asterisk 1.8.20.0. Currently we have a setup for inbound calls like that: Inbound call > Announcement (30 sec) > IVR (...
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Asterisk PBX Manager not saving changes

We are implementing Asterisk 11.4 and using the native PostgreSQL database to store call data. So far, we have not been able to change the admin password through the PBX manager. The password gets ...
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Switch from E1 to Analog PSTN line on IP-PBX failure

I have two Asterisk servers, E1(PRI) Gateway Asterisk 1.4 with E1(PRI) card IP-PBX Asterisk 1.8 on VM Server When my e1 gateway is failed (ex. shutdown, no power...), PSTN automatically switch to ...
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Starting SecAstQFSFileEngine::open: No file name specified - error when starting secast

I'm receiving the below error when starting secast service secast start Starting SecAstQFSFileEngine::open: No file name specified [FAILED] Has anyone run into this issue before? I'm running CentOS ...
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Blocking geographic cities from accessing Asterisk using Secast

I am using Secast for intrusion protection on my Asterisk PBX. It’s working great, and I now want to start blocking specific geographic regions. My system is getting hammered from Ramallah ...
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C3X Polling Strategy - Fewest Answered

I've been playing around with a 3CX PBX system and have been trying to alter a call queue so that members of staff that have had the fewest calls will get called first. So I have set the call queues ...
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Asterisk, How can i play an audio file [closed]

Here is the dial plan [testInComingCalls] exten => s,1,Answer exten => 30953025,1,Dial(SIP/20000,20) I would like to play an audio file as soon as somebody answered the call (Welcome Message)..
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DTMF Detection in Asterisk PBX

I have an asterisk server as IP PBX. I have created an Auto-Attendant for my company and asking users to enter desired extensions when they call. The problem is that when the message is playing and ...
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3 answers
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How to move a Panasonic kx-t7633 Phone

We have Panasonic kx-T7633 phones and I need to move them across office how do I activate or fix the connection after I move the phone Looked at the manual but no answers and googled it as well any ...
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Advice respect an architecture PBX [closed]

i'm new on the world of PBX and network infrastructure. I worked in a model and wanted to know how feasible it is to implement it in a production environment. The requirement is to create a solution ...
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Limiting Bandwidth per port Cisco router

Let's say a businessses purchases a sufficently fast internet connection (such as 60Mbps cable connection) and VOIP quality is pretty good for the most part. The only thing I would be worried about ...
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Asterisk PBX on linux (CENTOS 6.2) hangs up calls that are being recorded after 33 seconds

i have an issue with my calls being made via ASTERISK PBX running on linux CENTOS 6.2. The use case is that a call is triggered from /var/spool/asterisk/outbound/ The caller is dialed The dialplan ...
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What is the best option for SIP service redundancy?

From your practical experience, what is the best option for SIP service redundancy? a) one SRV record pointing to a hostname which has two A records b) two SRV records pointing to two different ...
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PBXact Behind ASA 5505 nat udp range 10000-20000

I'm installing a new pbxact phone server onsite and I've ran into an issue using a 3rd party sip, I've got the acl's for the traffic created and the traffic is allowed, I'm assuming i need to create a ...
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Shoretel Upgrade Path

I currently have a Shoretel Server running Server 2003 x32 as a virtual machine paired with a ShoreGear 90 switch and another unused switch of the same model being reserved for manual failover. I am ...
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Provisioning SIP Phones over the internet

I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get ...