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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.

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32 views

Firewalld rich rule doesn't immediately block IP address

I have Asterisk installed on a CentOS 7 server that uses the Firewalld firewall. To prevent brute-force attacks, I am setting up Fail2ban to ban IP addresses that enter incorrect endpoint credentials. ...
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10 views

How to run sh file in Asterisk dial plan

How do i run sh file from Asterisk dial? tar -zcvf /backup/"backup_chan_folder_-$(date +"%Y-%m-%d").zip" `/root/chan/*.*` extention conf file exten => 2004,1,Playback(restarting) exten => ...
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20 views

How do I add static video to a call?

Problem: my conferencing software's SIP bridge only supports calls with audio and video. Now we can all agree that this is broken – in fact I already emailed my opinion about this to their support – ...
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90 views

One way audio with Asterisk 15 and a WebRTC client

I have two ways audio when calling from a WebRTC client to a mobile phone (through Asterisk), but when I call from the mobile phone to the WebRTC client the client rings, the call is established but ...
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32 views

programmatically initiating a call and connecting it to a conference on asterisk

I'm looking for a way to "script" a sequence where the following things happen in order Initiate a call to a predefined number When the call is answered, connect the call to a conference room. Call ...
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1answer
85 views

Asterisk nat=force_rport,comedia with routing

I have a setup where the phones are in 2 subnetwork, but only routing used between them not real NAT. 10.4.0.40 < Asterisk PBX 10.4.0.9 < Phone 1 10.1.0.66 < Phone 2 Phone 1 and the PBX ...
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1answer
210 views

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 ...
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1answer
172 views

Asterisk 13.21.cert3 error loading res_rtp_asterisk.so module

We are upgrading (or trying) our asterisk servers from version 13.13.cert9 to version 13.21.cert3 and we are facing some issues. This are the steps followed to build it : Download and untar the ...
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1answer
25 views

Connecting a Fujitsu SS-170A phone to Asterisk

I have recently acquired a Fujitsu SS-170A VoIP phone in order to connect it to my Asterisk server. First things first, I am able to perfectly use softphones in the same network I am setting up the ...
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95 views

in process_sdp: Ignoring video stream offer because port number is zero (ONE WAY)

I have found dead ends for this problem on different forums and sites, it would be time to solve it for everyone's good. Given 2 devices: A cisco 894x IP phone with h264 support B softphone software ...
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16 views

Asterisk server is not trying to register to trunk provider

I'm using an asterisk packaged from questblue, it's a gui on asterisk on centos. No matter what I try, in gui or adding register=user:pass@provider.example.dom/ext register => user:pass@...
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54 views

Using Asterisk as a gateway to Provider

I have 2 sip servers on different LANs. Freeswitch and another is Asterisk. Asterisk sits on a VPN with a provider and who has provided DIDs. All users register on Freeswitch. How can I route calls ...
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1answer
102 views

Puppet Firewall string rate limiting not working for Asterisk

I have an Ubuntu 16.04 server and using Puppet 5 with Asterisk 14.6.0 running. I'm trying to implement the string rate limiting rules as described in the link below but it's not working. The CLI ...
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1answer
197 views

Asterisk PJSIP Jitter Buffer

I'm currently using FreePBX which has GUI settings to set Jitter Buffer for SIP, but not PJSIP. We have around 90 remote extensions using PJSIP and i would like to enable the Jitter Buffer for all as ...
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113 views

Asterisk/SIP - server with two NICs

We are trying to setup our Asterisk server with two NICs on it. One NIC is used to connect to a local network (which has access to internet and other sip trunks over internet) and other NIC is a ...
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30 views

Queue MOH randomly unavailable

I'm setting up an Asterisk box, but I'm facing a problem with my queue's MOH : I hear it barely 1 time through 2 or 3, and I can't figure out why. Here is my sip.conf : [general] allowguest=no ...
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1answer
59 views

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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1answer
332 views

asterisk playback sound when someone answers

This is an asterisk CLI question. I want to be able to play a sound byte with the asterisk "originate" command, but only when someone answers, like: channel originate *insertoutboundlinehere* ...
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45 views

Why asterisk listens on port 50360?

When I run: netstat -tulpn | grep asterisk I see the following: tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN 12337/asterisk tcp 0 0 0.0.0.0:5060 ...
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1answer
174 views

Asterisk JITTERBUFFER function details

Asterisk v15 documentation states that JITTERBUFFER function is used to "add a Jitterbuffer to the Read side of the channel". Take this dialplan example: [from-pstn] exten => 1234,1,Set(...
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146 views

Asterisk: Configure the Server header field

How do I modify the Server header field in responses by Asterisk? By default, it includes the release number and the package name: Server: Asterisk PBX 13.14.1~dfsg-2+deb9u3 Server: Asterisk PBX 11....
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1answer
356 views

Disable IAX module on asterisk. Only allow SIP

I wanted to see on what ports my ubuntu machine was listening on; therefore, I ran the command netstat -tulpn. When I ran that command I found out that I was listening on port 4569. That is being ...
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511 views

When secure calling with asterisk (SRTP), why are client certificates needed for SIP devices?

I just added security to Asterisk by following this tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial#SecureCallingTutorial-Keys Note that asterisk does not install by ...
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1answer
254 views

Asterisk uses the wrong variables from sip.conf

I completed this tutorial in order to make secure calls with asterisk. Secure Calling Tutorial | Asterisk Project Wiki I am running asterisk version 13.19.2 on Ubuntu version 16 (debian) and as soon ...
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474 views

How to enable asterisk Call Completion on Busy Subscriber (CCBS)?

I cannot enable Call Completion on Busy Subscriber (CCBS) on asterisk, witch is part of the Call Completion Supplementary Services (CCSS), as is the Call Completion on No Response (CCNR) feature. ...
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236 views

rSyslog - Disable to push custom logs to messages

I sending my Asterisk's logs to Graylog. (its working fine) But the rSyslog write every asterisk's log to /var/log/messages not to the /var/log/asterisk/full. Asterisk logger.conf: [general] ;...
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1answer
280 views

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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2answers
506 views

Causes of RTP jitter at the server

Investigating some call quality issues (0.5 – 1 second dead spots in calls) I took a packet capture of a phone call between two extensions on the same PBX. Since I was capturing from the PBX, I was ...
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3answers
2k views

Asterisk compilation issue

I've downloaded source code of Asterisk from http://downloads.asterisk.org/pub/telephony/asterisk/ I'm getting error while compiling this from source code in Ubuntu 16.04.1. Please suggest ...
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1answer
277 views

“register” to asterisk with any password

I have set up an asterisk server on a non-important VPS to test something; there is no sip trunk, so it can't call anywhere. I see a lot of attempts to register; I'd like to allow them to register to ...
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1answer
341 views

FreePBX - How to re-write forwarded “follow-me” caller ID?

Sam calls Julie's direct dial line and because she's not at her desk (and because of a follow me rule) the call is forwarded to her mobile. FreePBX forwards the call through the correct outbound ...
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1answer
127 views

Asterisk (VOIP) port forwarding on TP-LINK [closed]

I have local asterisk server on UBUNTU 16.04. It's address 192.168.0.101:5060 and I would like to be allowed to talk with other users when I am outside local network. I tried THIS but it doesn't work. ...
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1answer
339 views

“channel allocation failed: refusing due to active shutdown” warning

Our phones are down with the warning above when I try to run command show channels. Is a core restart was done, how long before phones back up? Or is there something else that needs to be done to ...
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1answer
271 views

Asterisk / FreePBX: Calls to internal extensions require users to press Dial

Using FreePBX 14 and the Cisco SPA504g phone, when my users dial outbound US number (e.g., 15555551212) the calls automatically dial. However, when the dial 555 for an internal extension, they have to ...
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438 views

Asterisk Confbridge conference. How to avoid eavesdroppers?

Trivia: I'm trying to implement a conference call management functionality on my Elastix server. The original problem was in the Elastix's unability to invite all participants in the conference via ...
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1answer
646 views

CentOS 7 Asterisk + FreePBX Install:

I attempted to run a ./install -n on my FreePBX install and got: ===== Checking if Asterisk is running and we can talk to it as the 'asterisk' user...Error! Could not determine Asterisk version (got: ...
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2answers
198 views

puppet iptables string rate limiting

Does puppet firewall support the string module? I need help to implement the following iptables rules in puppet but can't find any documentation or examples on how to do this. iptables -I INPUT -p ...
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1answer
176 views

New Asterisk 13 & 14 missing Application “Read()”

I just upgraded from Asterisk 11 (Elastix) where this script was working to Asterisk 14. I am getting a slew of errors that are surprising. It appears problem is that there are some "default" ...
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1answer
348 views

Asterisk / FreePBX: How to differentiate incoming calls?

I have an FreePBX/asterisk server setup with two incoming trunk lines with different DIDs, e.g., 555-1111 and 555-2222. However, the lines represent two different entities operated from the same ...
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390 views

Vicidial GoAutodial - move MySQL database to another server

I have installed GoAutodial version 3.3 (this is essentially an automated installer for the Vicidial autodialer which uses Asterisk to place calls). It automatically installs a MySQL database on the ...
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1answer
291 views

Connect one SIP phone to multiple Asterisk Servers

I am working on a project where there are multiple apartment complexes that share the same reception desk, it's just one of the buildings. There are VOIP devices in those apartments that have to be ...
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1answer
256 views

Asterisk: AMI Mute in conference doesn't mute DTMF

I am on Asterisk certified/13.13-cert3 and running into this problem. I have a dialplan like so: [extensions] exten => _X.,1,noop same => n,ConfBridge(123) I have a conference profile with ...
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1answer
436 views

MixMonitor applied to every extention in a context

I'm trying to setup call recording in Asterisk applied to a whole context. extensions.conf looks like this: [office] exten => _068.,1,Dial(SIP/sip_trunk_1/${EXTEN}) exten => _073.,1,Dial(...
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1answer
45 views

Can Asterisk passwords be obtained by a malicious server?

To make a long story short, a client phone was compromised and used to make illicit calls. An investigation revealed a port forwarding entry to the phone's web UI, which is "protected" by a 6 digit ...
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2answers
587 views

Choose outgoing trunk in custom context FreePBX

I need to be able to choose some trunk based on result of some SQL query, while making outbound call, i spent many time google some answers, but i cant find any. I know i can use outbound routes in ...
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1answer
374 views

Allow inbound and outbound calls on same asterisk (number not registered)

I have a particular case where carrier (not provider), give me calls without having to register. On carrier side, numbers have an associated IP which is used to route any call. FOr outgoing call, I ...
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2answers
3k views

Asterisk redirect calls to external number

We have a phone system created on pure Asterisk (no web GUI) and there are the usual Day/Night mode. In the day mode it calls a ring group. However we have a call centre that we use to pick up our ...
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1answer
54 views

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 ...
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1answer
54 views

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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1answer
478 views

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 ...