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

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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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94 views

How to set Mitel 5224 ringer volume in configuration?

We have several Mitel 5224 SIP phones, but can't seem to find out to make the ringer volume setting persist after it reboots. Can the ringer volume can be set somewhere in the XML configuration file?
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239 views

Change the caller id from my asterisk server

I successfully set up an asterisk server. When people call my asterisk server via PSTN, the server will place another PSTN call to my phone at 33344455555. When I receive the call, my phone shows ...
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478 views

Asterisk ${CALLERID(name)

I am working with asterisk (Asterisk 1.8.11.0), freepbx (2.x) and was building a dialplan (extension_custom.conf). I wanted to get caller Name from database which i have stored for example. CLI> ...
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441 views

How to forward a call to SIP server

I have a SIP server(Asterisk) I have configured a DID for outbound and inbound. My asterisk server has freepbx installed to manage it. My requirement here is, when anyone calls to this DID call should ...
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35 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....
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70 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 ...
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19 views

Having trouble with one way adio on an asterisk sip phone

I have an asterisk server (Elastix on CentOS 7) that I am currently running in Amazon Web Services. The server works great and clients can connect without issue using a stun server. However I have ...
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25 views

Voicemail location with two Asterisk servers connected by Follow-me

Here's our current setup: We have two Asterisk servers in different buildings setup and connected by IAX. Call them ServerA and ServerB. We have several DID trunks coming into Server A, where our ...
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46 views

Asterisk redirect incoming call to last number that called it

I would like to know how to redirect an incoming call to the extension that called it first in Asterisk. I have not found a way to do it, and I haven't found any Asterisk variable that contain a value ...
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80 views

Issue with Asterisk on Google compute engine (GCE)

I installed and configured Asterisk in Google compute engine (GCE). I am facing the below issues: 1) Unable to login to Asterisk CLI by using asterisk -rvvvvv command (But, I am able to login using ...
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565 views

Asterisk dialog destruction on attended transfer timeout

Steps to replicate: Alice receives incoming call from Eve (either from external line or another extension, doesn't matter). Alice makes attended transfer to Bob and hangs up before Bob answers. Bob ...
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238 views

How to know what number was dialled on incoming call

The scenario In the UK. I have one (BT openreach) phone line coming into a Cisco SPA232D. That phone line is ringable on a number, let's say 01235AAAAAA. I have a virtual number, 01235BBBBBB, which ...
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96 views

Asterisk one way audio detection

I have some calls have one way audio. I tried to write ${CHANNEL(rtpqos,audio,all) to CDR and think what if themssrc=0 than no audio, but it's not true. Is any way to detect one way audio? ssrc=...
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483 views

sip register success on one host but fail on another host in same local network

I acquired antisip account and setup asterisk (lets say PBX-A) to register. It worked fine. However, I attempted to do so with another asterisk (PBX-B). No luck this time. PBX-A and PBX-B are on the ...
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2k views

Vicidial in Autodial does not connect correctly dongle phone call to agent conference

I want to make work Vicidial with chan_dongle Asterisk module to use Huawei USB modems as GSM gateways (trunks/carriers) for asterisk. In manual mode everything works. Problem appears only if ...
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217 views

Werbrtc2sip crashing intermittently. Aborted (core dumped)

I am having issues with making webrtc2sip actually work. Webrtc2sip is now getting crashed intermittently during a call. I have seen this even when only one call is active in a queue. By the way, my ...
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260 views

Connect Cisco 7975 VOIP phone to two separate Asterisk PBX's

I have a Cisco 7975 registered on an Asterisk SIP server, and I want to register a second line on a second (different IP) Asterisk SIP server. I tried to do this with firmware version 8.5.3 but it ...
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370 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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2k views

Error raised during calls in asterisk server

Today my asterisk could not receive calls for a while. when I checked the asterisk logs, the error below was raising during this period. Purely numeric hostname (1002), and not a peer--rejecting! ...
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214 views

How can I use the FLASH key of my IP phone to transfer a call (atxfer)

My IP phone has a FLASH key. This key sends the DTMF of Flash [16 in decimal] (http://www.voip-info.org/wiki/view/SIP+DTMF+Signalling). How do I set the atxfer parameter in the features.conf?
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475 views

UDP Range Nat Translation port forwarding RPT Traffic to inside host PBX Cisco ASA 5505 8.2a

I have an Asterisk based PBX that I'm providing VoIP services with, My gateway is a Cisco ASA 5505 running 8.2 and I only have one public IP Available along with other services I provide with that IP. ...
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542 views

Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic (...
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359 views

Asterisk 1.8, Dedicated SDSL but SIP Provider Inbound Channel heavily delayed?

We have a dedicated Asterisk 1.8 server running as a front end SIP server for authenticated and internal users only, using SDSL 2MB/2MB, in the UK. The provider of our internet also is the provider of ...
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3k views

Configuring Hylafax to work with Asterisk over a single SIP provider

I'm setting up an Asterisk 1.8 based PBX, and as part of the requirements, I need to make provisions for sending and receiving faxes. The 'big picture' goal is installing a fax printer at each Windows ...
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172 views

asterisk Dial-plan?

I want to make a dial plan for asterisk to do this: for incoming-call check the caller-id and if caller id is equal with specific number (for example 666) then hangup that call.(this dial-plan also ...
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41 views

add presence for a remote user to a legacy telephone system?

we have a small call center that uses an old nortel phone system with analog lines. one of our sales people works from home so her calls do not go through the phone system. this creates a problem at ...
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318 views

How to send T.38 from a mac?

I'm trying to set up a fax-server on a macintosh. I have Hylafax, and we're going to use an internet FOIP fax provider (Haven't decided who yet, that may be another question). The problem is how to ...
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9 views

Asterisk: How to disable specific dahdi channel without reloading module?

My POTS line #2 (which is channel 2-1 on Dahdi group 0) is having noise issues, while the phone service provider is fixing that problem I would like to disable this channel temporarily. Is there a ...
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7 views

Outbound Rule not working AsteriskNow

I am trying to set up outbound routes for UK numbers (both landline and mobile). UK numbers follow this convention: 0[12378]000000000 However, I cannot make outbound calls on my AteriskNow server. I ...
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27 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 ...
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41 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 ...
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231 views

Asterisk internal call not routing correctly. Unable to retrieve PJSIP transport 'udp,tcp,ws,wss' for endpoint 'anonymous'

I am trying to figure out why my FXO adapter has suddenly stopped working, it has been a while since it was first configured, I had only changed an internal call timeout setting on the FXO adapter and ...
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38 views

Can't route traffic from Cisco to Asterisk over port 5060

My VOiP provider has informed me that I must route incoming calls from my Cisco router to my Asterisk server to port 5060. I can ping my Asterisk server from my Cisco router fine, there's a direct ...
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26 views

OPENVPN ipp.txt file assigining duplicate ip addresses

I am having an issue with duplicate entries for hosts in the ipp.txt file where the same host CN has been added multiple times with different ip addresses. For some or other reason some of my openvpn ...
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82 views

DTMF Tone issue with Elastix

I am very new to Elastix and have just gone through the setup of a new system on Elastix 2.5.0-8, I have Elastix in the middle of 2 sip trunks, one is the service provider and the other is a 3rd Party ...
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45 views

What is needed to enable a passwordless SIP connection in Asterisk to accept inbound calls?

I have an Asterisk system with multiple users. Until now, where a user required the ability to place calls, we provided them with a SIP/IAX account with a username and password, plus a custom routing ...
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43 views

How to connect extensions of remote VOIP over remote STUN server?

Image of Architecture I am trying to setup I have configured the Asterisk server in a docker image and it's running, I have also configured the STUN/TURN server and it's also running on the same ...
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62 views

Enable Google Authenticator to send codes via SMS with OpenVPN

Setup: Using OpenVPN in single-server<->multi-client configuration. Running google authenticator PAM module in order to enforce 2 factor authentication. Problem: Some employees don't have a ...
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142 views

Vodafone SIP trunking

I came here to ask if anyone of you has experience with Vodafone Business SIP trunking in Germany or in general. Vodafone offer (german). The reason I ask is because I'm planning on setting up ...
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108 views

Configure Elastix to show incoming caller name from predefined list

I have configured call center in Elastix. My task is to show caller name from predefined list of contacts when incoming call starts. Right now I have only numbers under Source column in Monitoring ...
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203 views

Asterisk recieves 403 on every outbound call after 5 minutes delay

I have a home server in my local network with Asterisk installed. It has been working without any problem for several month. I have just changed my router (ZyXel Keenetic Lite -> LP-LINK TL-R480T+) ...
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121 views

Calls on new telephony Server (Vicidial) being “dropped” after 30 minutes being INCALL?

Hello everybody! Im in a weird situation with my Vicidial Cluster with the SIP Account 1. I have actually 3 Servers: 192.168.1.200 (All-In-One Server) 192.168.1.201 (Telephony 1) 192.168.1.202 (...
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17 views

How to recognize conference calls from individual simultaneous calls When we have Asterisk user agents?

I want to know how I can distinguish between conference call from some independent simultaneous calls when we have Asterisk agents? RFC 4353 says that in conference calls we have a call member with a "...
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310 views

Asterisk SMS with GOIP

I have a GOIP 1 channel gateway and following this guys tuorial: https://github.com/tangaza/Tangaza/wiki/Connecting-a-GoIP-GSM-VoIP-gateway-to-an-Asterisk-AGI For me receiving SMS-es only works if ...
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206 views

freeswitch call getting abandoned

i have a freeswitch box that successfully handles incoming calls to a certain DID number, but when i call in from another DID number (which happens to be from Asterisk) i get a message that the call ...
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107 views

OpenSSL conversion from Windows CA for Asterik / Switchvox X.509 PEM Format

I'm trying to sign our switchvox PBX web certificate with our windows AD CA. SV Keeps telling me the certificate is invalid and must be in PEM format. Can anyone see what i am doing wrong? I Think ...
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258 views

Asterisk SIP Retransmission timeout after a month of server uptime

This Asterisk server runs fine behind NAT with a dozen of SIP registrations and the call volume of a few hundred calls per month. But then, after about a month of uptime, out of the blue the SIP ...
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2k views

Asterisk 11.* TLS Configuration

I have installed asterisk 11.12.0 on CentOS (64-bit). When enabling the TLS support i get the following error WARNING[7620]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! i tried ...
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200 views

Using IAX2 to call from one Asterisk server to another

I have two asterisk servers serverA and serverB . I am trying to place a call from serverA to use serverB using IAX2. I have defined a user on serverB in iax.conf. When making the call this is what I ...