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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How does buffering work in a chain of Asterisk servers?

I have two Asterisk servers in a chain like this: SIP client (G.729) <-SIP-> Asterisk 1 (converting G.729 to G.711) <-IAX2-> Asterisk 2 -> terminated to analog line. When no buffering was ...
Alexander Shcheblikin's user avatar
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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 ...
matteo nunziati's user avatar
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FAILED status in Elastix Reports

Please i want to know what is failed status means in the CDR report in Elastix, i to know if failed status mean that the call is dropped because the agent wasn't registered or the voice mail is full ...
ahmed saad's user avatar
2 votes
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How can I tell asterisk to initialise each music on hold application when needed rather than constantly

I'm running a custom MoH application for a large number of users under an asterisk 11 box. The configuration is as follows: [customer-100] mode=custom application=/usr/bin/ruby /path/to/script.rb 100 ...
Andrew White's user avatar
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1 answer
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SIP registration timed out

I am facing trouble in registering asterisk to SIP trunk. When I do >sip show registry, it shows SIP request is send but never gets response back. I cannot even ping sip.flowroute.com. My firewall ...
bluewhale's user avatar
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4 answers
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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:[email protected]....
bluewhale's user avatar
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2 answers
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How can I setup Asterisk as a Voicemail server for CallManager?

I have an existing Cisco CallManager 8.5 environment which services users needing voicemail, but I have no Unity licenses. How can I use Asterisk to service voicemail?
Mitch's user avatar
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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 ...
Tono Nam's user avatar
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"Generico" networking issues: how to investigate

Recently I'm experiencing networking issues in my LAN. I have an Ubuntu server doing PDC, mailserver and Asterisk. There's KVM on it, which runs a Windows VM and pfSense firewall. From time to time ...
Maxxer's user avatar
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2 votes
2 answers
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UFW (firewall) not blocking UDP right away

I have a PBX (phone system called Astersik) that works with udp and tcp. Sometimes I get invalid request from some IP addresses that I will like to block. I cannot block those requests until I reboot ...
Tono Nam's user avatar
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Can Asterisk's phoneprov module be used securely?

I'm using Asterisk 13.1.0 as packaged by Ubuntu Server 16.04 to run a pure-VoIP phone system. Asterisk has a module – phoneprov – that allows it to template out configuration files for ...
Sam Hanes's user avatar
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Asterisk times out and terminates connection after 6400ms due to 'no response'

I have a SIP trunk set up with Twilio for outbound calls. Twilio-FreePBX and then my test device is the simple X-Lite from CounterPath. I can make an outgoing call from X-Lite. My cell phone rings ...
Max Phillips's user avatar
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146 views

Asterisk CLI Verbose[?number?]

In my CLI, I have entries like VERBOSE[16933] logger.c: WARNING[8556] chan_sip.c: VERBOSE[8556] logger.c: VERBOSE[16932] logger.c: What is the significance of the number with the "[]" brackets?
jerryrig's user avatar
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Asterisk client behind NAT - UDP registration does not work

I have some clients connected to my Asterisk server behind a NAT device. When using TCP everything works OK. If I change the client and server config to use UDP (from transport=tcp to transport=udp,...
fileinsert's user avatar
5 votes
1 answer
6k views

How to connect an existing phone-line to asterisk server

At office I am considering setting up an asterisk server to that we can route calls that come into the office internally so that they reach the correct person and to implement a menu system as well. ...
Cameron Bartlett's user avatar
1 vote
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40 views

Detect files behind I/O wait issue

We are running Asterisk 1.8.22.0 which uses NFS for sound files (for IVR's), voice mails etc. Every so often the load average gets real high. Anytime that is we see the I/O wait in top go over 7%. The ...
Dovid Bender's user avatar
-2 votes
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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 ...
real_sm's user avatar
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asterisk cluster with shared database

I want to build an asterisk cluster from 3 asterisk (13.x) nodes, one on US, one in Europe and one is Asia. Right now I have the 3 servers. Asterisk is using realtime infrastructure for sip/iax users,...
Videanu Adrian's user avatar
2 votes
6 answers
5k views

Asterisk ghost calls

Problem We have an Asterisk server hosted externally. On four locations we've gotten ghost calls. These are calls with different numbers like 1000, 9999 or 6060. We don't use these numbers, not even ...
SPRBRN's user avatar
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CPU load too high after upgrade from 1.8 to 11.23.1

Feel free to shoot me down in case there is more data that i can provide. I have gone through some blogs on serverfault and other blogs and gathered as much info as I can. I want to track/fix cause ...
Krishnam's user avatar
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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. ...
nameLess's user avatar
0 votes
3 answers
546 views

Polycom SoundPoint IP 4000 won't power on. How to isolate power supply?

I have a Polycom SoundPoint IP 4000 SIP conference phone that has sat unused in a box for a number of years. Today, I pulled it out of the box and powered it on with the conference power supply only ...
Steven Sokulski's user avatar
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2 answers
1k views

Asterisk: music on hold stuttering

We're having issues with music on hold (moh). Occasionally (maybe every second time) a caller is put on hold and moh is played, the music is stuttering/stammering (in various degrees - sometimes you ...
Peter's user avatar
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No sound in XLite softphone calling Asterisk PJSIP channel bound to callcentric

When calling from an XLite softphone to a Callcentric number which has an Asterisk PJSIP channel registered, we cannot hear anything at all on the softphone (though the call is indeed established). ...
Santiago Palladino's user avatar
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1 answer
1k views

Configure asterisk for blink and linphone

I have setup an Asterisk server (14.0.2) on Ubuntu 14.04. I can get sound from Twilio using ulaw and from Zoiper (no STUN or ICE). In each case the asterisk server plays gsm files. I can't get any ...
markhorrocks's user avatar
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1 answer
663 views

Asterisk - can't dial if hints are used

I'm trying to learn Asterisk by following the Definitive Guide. I'm trying to achieve the following: Extensions 7001 7002 7003 ..each with hints so my IP phone handsets can show busy lights. I ...
fistameeny's user avatar
2 votes
1 answer
988 views

Unable to dial SIP phone connected to 3CX system via Asterisk

So i have a rather complicated issue I am trying to connect my Asterisk with a client 3cx phone system So there is a sip phone connected to 3cx system with extension 351 On my Asterisk I have added ...
Maciej Cygan's user avatar
1 vote
4 answers
503 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 ...
Kramer786's user avatar
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Why is my linksys spa not recognizing hangup?

I have a Linksys SPA which is connected to my one asterisk server. When I receive an incoming call, the PSTN line of the Linksys is forwarding the call to my server. However, when it's going to the ...
Loebstahhhh's user avatar
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1k views

Asterisk sslv3 alert handshake failure

I am using Ubuntu v14.04.3 LTS and Asterisk 13.3.2. When i try to call to my extension from a sipml5 client to just play a demo-congrats audio, my call gets disconnected instantly. When i check ...
Ijas Ahamed's user avatar
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1 answer
842 views

Connect Asterisk system to SIP provider with Asterisk dynamic realtime

Is it possible to connect an Asterisk system to a SIP provider with Asterisk dynamic realtime (ARA)? In other words, is it possible to configure a table (or perhaps view) such as sippeers for family ...
Drux's user avatar
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Use SQL views instead of tables with Asterisk dynamic realtime

I had hoped that I would be able to use an SQL view instead of a table to provide e.g. iaxusers and iaxpeers to Asterisk via dynamic realtime (ARA). I have just noticed that iaxpeers apparently ...
Drux's user avatar
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Check Asterisk dynamic realtime (ARA) configuration with CLI

After configuring Asterisk dynamic realtime by adding the following lines to /etc/asterisk/extconfig.conf, is there any way of checking the resulting settings on the level of Asterisk CLI? [settings] ...
Drux's user avatar
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1 vote
1 answer
2k views

Asterisk record outgoing calls

i am trying to recotd outgoing calls on asterisk and I always get empty audio file. Can understand why :( Part of [outgoing] to test calls exten => 8916XXXXXXX,1,NoOp() exten => 8916XXXXXXX,n,...
Hikaru's user avatar
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1 answer
669 views

Asterisk channel variable set before dialplan executes?

The backstory is that I am trying to debug an issue I am having with FreePBX currently. The channel variable REC_STATUS is set to RECORDING when it shouldn't be. After quite a bit of troubleshooting ...
YWCA Hello's user avatar
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0 answers
127 views

ISP SIP account on own Asterisk machine

I'm new to VoIP and SIP, have experience in Linux and servers in general just not SIP servers and such. I was curious about experimenting a little with SIP and VoIP. So I ordered a SIP account from ...
Ecaz's user avatar
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2 votes
1 answer
5k views

Zoiper softphone cannot register to Asterisk with IAX2: cause connection refused, cause code 29

I am trying to configure a Zoiper softphone (iOS app version) so that it can connect to my Asterisk PBX. The Zoiper configuration is as follows: Account name: Account1 Server: <ip-address-of-my-...
Drux's user avatar
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3 votes
2 answers
4k views

Logs concerning failed IAX2 registration attempt missing on Asterisk server

I am trying to register a VoIP softphone (Zoiper app for iOS) with an Asterisk installation via IAX2. At present, this results in an error message "Registration Refused (29)" on the side of the ...
Drux's user avatar
  • 696
1 vote
1 answer
1k views

Attaching to Kubernetes container (for Asterisk) in GKE leads to endless cycle of attach/detach

I have managed to deploy Asterisk to Google Container Engine (GKE) currently to the extent that a VoIP softphone can register. I'd now like to attach to Asterisk's container. kubectl attach -it <...
Drux's user avatar
  • 696
2 votes
2 answers
941 views

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. ...
bloopiebloopie's user avatar
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2 answers
235 views

What are standard ways to pass text from asterisk to a IPPhone/Softphone user

I have some leads to whom I want to make outbound calls. Along with the phone number, I have additional information like Name, Query Source and Query Category. "John W. Lead", "+43 780 004711", "...
Pranab's user avatar
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Delay in outbound & inbound calls for ip phones

I have an issue with outgoing calls and inbound calls on elastix 2.6 some time i can make calls from ipphoe but its take more than 10 to 25 seconds Or cannot make calls Incoming Some time phone ...
Joseph's user avatar
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4 votes
1 answer
8k views

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 sip.conf [201] user=201 secret=123456 type=...
Agent69's user avatar
  • 141
1 vote
1 answer
491 views

Asterisk automatic installation

I recently started a project which I need to automate the installation of Asterisk servers for my customers in AWS infrastructure. Customers already own AWS infrastructure. I was thinking the ...
gogasca's user avatar
  • 343
1 vote
1 answer
5k views

Asterisk's "dialplan reload" and "help" commands are not found

I've installed Asterisk 13(current) on CentOS7(without DAHDI/LIBPRI). It seems to work. I was following instructional videos, but at some point these use dialplan reload and help commands. These ...
Whoa's user avatar
  • 11
1 vote
1 answer
993 views

Four node asterisk cluster

I have a question regarding Asterisk servers. I have a client that have two locations connected with fiber optic. On both locations they have PRI ISDN-s. On one location there are 4 PRI ISDN, and on ...
Branko Markovic's user avatar
0 votes
2 answers
172 views

ISDN BRI vs SIP Trunks over EDIA

I am installing Asterisk in a small office with just 4 lines and 8 Extensions. I have two choices from my local telco (Fairpoint): 1) Old School ISDN BRI lines which I would connect to Asterisk ...
iewebguy's user avatar
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1 vote
1 answer
8k views

Asterisk: How to enable the HTTPS mini server?

I am attempting to enable the HTTPS mini server in Asterisk with little luck. Here is my http.conf: [general] enabled=yes bindaddr=192.168.254.26 bindport=8088 tlsenable=yes tlsbindaddr=192.168.254....
MacroMan's user avatar
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0 answers
843 views

Asterisk/FreePBX: how to reject all incoming calls after few seconds?

I have 3 dumb GSM relays that control the gates. If a person whose number is in device's address book calls, gate opens. Call gets rejected (in my country calls that didn't connect or lasted less than ...
Alexander's user avatar
1 vote
0 answers
997 views

send fax with Elastix and Hylafax

I need help getting around "Unknown problem" while I am trying to send faxes using Hylafax and Elastix. I have an elastix 4.0 inside a VM, calling outside and inside the box is working great, using ...
cornel's user avatar
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