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Questions tagged [asterisk]

Asterisk is software that enables a server to act as an IP PBX system, VoIP gateway, conference server, and more.

see

http://www.asterisk.org/

and

http://en.wikipedia.org/wiki/Asterisk_(PBX)

586 questions
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Asterisk open source phone systems - determining which 800 phone number was used?

I have an Asterisk open source phone system. My business will have many toll free 800 phone numbers, they all go to the phone system. When a client calls us, is it possible to know which 800 phone number they used? Maybe somewhere in the call…
davidjhp
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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 arrive in the Cisco logs (debug vpm sig). We see the…
jonathanjo
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What bandwidth would be required for 55 VOIP Lines And what type of Internet connection

In a call centre environment, what bandwidth would be required for 55 lines? This would be on SIP protocol, would we be best to use G729 codec as i know that our sip provider supports this? What type of internet connection would be best for such a…
Gary B2312321321
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Register asterisk to sip trunk

I want to register my asterisk server to a SIP trunk. I have added following piece of code in my sip.conf and extensions.conf sip.conf [general] register => myusername:mypassword@sip.flowroute.com allow=ulaw [flowroute] ; keep…
bluewhale
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Call from '201' to extension '202' rejected because extension not found in context 'test'

I have this code on my extensions.conf [test] exten => 20,1,Answer() exten => 20,n,Playback(hello-world) exten => 20,n,Hangup() and this is my…
Agent69
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Redirect Call Back Out To PSTN

I'm wondering if it's possible on a VoIP system (specifically Asterisk) over SIP trunks to redirect a call back out to the PSTN to no longer be routed through the PBX. What I'd like to do is see about brokering connections between two 10 digit…
StrangeWill
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Asterisk behind NAT sets wrong Contact Header

I'm using SIP with asterisk 13.1.0 behind a statically configured NAT. The servers private_ip differs from the public_ip, where I can reach it. I've already set these options in the sip.conf…
Lukas Bernhard
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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 (no sound, for extension dial only, 5 sec) > Queue (for…
user2838376
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Limit simultaneous call per user in Asterisk

I want to limit simultaneous calls per extensions in Asterisk for security reasons. For example when a user is on the call no body else would be able to make a call by that extension. How can I achieve this?
Pooya Yazdani
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Sending faxes from Asterisk (email to fax) --- is any feedback possible?

I am about to make some email to fax gateway --- some python program will make Asterisk call files and put them into some queue (files or db based). Then another program should take "call tasks" from queue and "feed" them to asterisk (honoring some…
brownian
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Registering SIP phone (X-Lite) to asterisk server (asteriskNow)

We currently have a school-project going in which we need to set up an PBX with Asterisk. Thus we have installed asteriskNow in a virtual environment (with virtualBox) on ubuntu 12.10. We tried the configuration by registering some SIP-Phones via…
kafman
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How do I secure my Asterisk server?

Our asterisk server was compromised. some calls were made to Asia countries last weekend. Thought we have improved our network configuration, we still want to determine how the intrusion was done, we think there are clues in our asterisk log…
SDReyes
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Make call with alternate provider if NOANSWER

I have two voip providers, one free an the other paid. The free provider only allows local calls to certain area codes, so I need to fall back to the the paid provider if a call fails. At the moment, I have the following context in my…
Alfero Chingono
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How to stop registration attempts on Asterisk

The main question: My Asterisk logs are littered with messages like these: [2012-05-29 15:53:49] NOTICE[5578] chan_sip.c: Registration from '' failed for '37.75.210.177' - No matching peer found [2012-05-29 15:53:50]…
Travesty3
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Asterisk 15-minute drops calls

There seems to be a hard limit on incoming calls set a certain way- a telephone engineer told me about this, but I don't know the specifics. When on a call, the moment the call hits 15 minutes it drops. Does anyone know the specifics of this…
chrism2671
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