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 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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677 views

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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903 views

Lync Server 2010 with Hosted VoIP PBX

We just deployed Lync Server 2010 in our organization and it is working great so far. The next step for us is to enable enterprise voice so that we can replace our telephones with service that is ...
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722 views

What is a fairly accepted endpoints to concurrent calls ratio for a PBX?

Hey guys! Not too positive about the relevance of this question to Server Fault, but I'll let you guys poke around at it anyway. :) I'm trying to figure out what an accepted ratio of endpoints to ...
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1answer
277 views

What's the correct way to step a Draytek 3510 IP PBX to avoid line echo?

I've setup a Draytek 3510 IP PBX with 10 VigorPhone 350 handsets. It's all connected through an unmanaged 48port gigabit switch - all structured cabling except patch leads are Cat6. Dhcp is handed out ...
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267 views

Microsoft Lync and Avaya PBX system

I am a complete beginner concercing Microsoft Lync and PBX systems, so please bear with me. I have set up a Lync standard server which is running fine. We can chat/audio & video conference on our ...
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1answer
369 views

Where can I get some real life CDR/SMDR/AMA logs from asterisk or ther IP based switch?

In order to develop some generic call accounting software I need some sample logs from IP based PBX. Where can I get some? - I need more than 100 records, preferably using both SIP and telephony ...
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46 views

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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44 views

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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193 views

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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1answer
256 views

Connect to Panasonic PBX from linux box

We are developing a project in a linux box and want to connect it with a existent Panasonic pbx, but it seems that there's no tool to do that, and the only way to get connected is developing yourself ...
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528 views

Connection problem between PC and PBX machine

Hi I encountered the error message “Error Opening Com Port” on PC. That PC is used for PBX System and PBX (Panasonic Digital Super Hybrid System KX-TD500 version 2.5) uses RS 232 serial interface for ...
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38 views

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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0answers
177 views

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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39 views

Asterisk SIP calls between remote L2L VPN sites

I am running into an issue trying to set up another remote L2L VPN site with VoIP running SIP and Asterisk. Each remote Site (site 1 and site 2) Can talk to the Asterisk server and any phones on the ...
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11 views

Nortsatar-plus MICS on pri converting to server based PBX

I currently have a Cisco router providing our PRI, Fax, and Data. PRI to a norstar plus MICS PBX (NT7B53FA). I want to basically switch our norstar KSU for a server (2008 r2 standard or I can change ...
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62 views

Combining Elastix And Zultys PABX

have anyone integrated Zultys MX250 with Elastix? I got some problem with it. Before, my PABX is Elastix, now i want to upgrade using MX250 but keep Elastix as gateway for MX250. But i can't call out ...
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203 views

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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75 views

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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71 views

Connect Cisco IP phone on two Asterisk sip servers

I have a cisco ip phone with sip firmware v9.3.1. I want to connect two lines on two seperate Asterisk sip servers. In section i can only define USECALLMANAGER (in node) and not the IP of an ...
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35 views

Routing SIP client thru WAN router

I may ask again: I have following setup: SIP Client on LAN can make calls, etc thru PBX (is a closed device). I want the same for the remote SIP client. Can I configure all in the WAN router ? May ...