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 ignoring sip_domain part of URI

I am using Asterisk version 11.5. My problem is the following: I literally cannot make any outbound call. That's because, whatever I dial in any client, the sip_domain part gets ignored, it is ...
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1answer
48 views

Advice respect an architecture PBX [closed]

i'm new on the world of PBX and network infrastructure. I worked in a model and wanted to know how feasible it is to implement it in a production environment. The requirement is to create a solution ...
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92 views

How do I set the current working directory of Asterisk?

I need to be able to collect core dumps, but since Asterisk's CWD is / (so claims procfs), it'll never be able to write them. I've confirmed my suspicions by allowing world write to / and SIGABRT the ...
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615 views

Avaya-Asterisk call drops immediatly via SIP (sip:anonymous@anonymous.invalid)

I'm trying to call from Avaya to Asterisk (11.5.0) in SIP. Avaya calls to Asterisk, then Asterisk rings my softphone but when I answer the call (in the softphone), call drops immediately. This is the ...
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3answers
146 views

make web portal visible on two different vlans on same interface

We have recently set up a new Aterisk pbx server based on our previous pbx (also Asterisk) which was installed by an external company. We've kept the configuration of the server the same, all we did ...
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256 views

Asterisk PBX on linux (CENTOS 6.2) hangs up calls that are being recorded after 33 seconds

i have an issue with my calls being made via ASTERISK PBX running on linux CENTOS 6.2. The use case is that a call is triggered from /var/spool/asterisk/outbound/ The caller is dialed The dialplan ...
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2answers
1k views

High Load, low CPU, Memory and Disk IO - Highend Server

This issue has been bugging me for the past several days having spent over 40 hours investigating this issue intensively. Effectively we run asterisk 1.4.42 which I understand is old, however is the ...
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1k views

Asterisk and websocket: “rejecting secure audio stream without encription details”

I'm trying this library using the test service: http://tryit.jssip.net/ When I register, using WS, not WSS, peer is registered succesfully When I make a call i got "rejecting secure audio stream ...
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75 views

Asterisk console output 'no button pressed' on no DTMF input

I've got an outbound survey program, here is an example of what the survey does from the dialplan. same => n(xtn_age),Read(AGE,rbp_age,2) same => n,Set(CDR(age)=${AGE}) same => ...
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2answers
2k views

How to install Asterisk with SRTP support?

I'm rather new to Asterisk, and I need my server to support WebRTC. As far as I know, Asterisk version in Asterisk Now is compiled without SRTP support, which is necessary for WebRTC. So, I try to ...
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213 views

using iptables to secure asterisk

Following the post at http://www.dslreports.com/forum/r26964868- I created some IP tables for my asterisk install. The server is behind a router and the router only had 5060 and the RTP ports ...
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413 views

Asterisk and SIP trunking, is this configuration possible?

I am new to asterisk and before I dive in, I just want to make sure that what I plan to do is possible/correct. My office will run an asterisk server and have both local and remote extensions. We ...
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2answers
1k views

Interpreting and using the Asterisk “timing test” command

Timing is very important for certain kinds of applications in Asterisk. If DAHDI is the timing source, the dahdi_test command can be used to check the timing provided by the DAHDI kernel module. If ...
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1answer
209 views

Asterisk: Forcing a sip peer to connect via ipv6?

I've got an asterisk server that connects to an upstream provider over a WAN. The upstream provider supports both IPv4 and IPv6 connectivity, and the asterisk server is behind a NAT. When asterisk ...
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1k views

Get details about “Remote UNIX connection disconnected”

I am getting lots of: -- Remote UNIX connection disconnected I would like to get more details about it. How do I get information about who is connecting, ip address, and more details?
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240 views

Monit detecting error, but not restarting asterisk

I am running Ubuntu 13.04 with the corresponding versions of monit (5.5-6) and asterisk (1.8.13.1). I have monit set up to watch my asterisk log file for disconnection to my SIP provider and restart ...
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306 views

can't seem to install dahdi on my server

I am trying to install the dahdi module on my server but it keeps failing when i run the make all command. See below: root@server:~/dahdi-linux-complete-2.6.2+2.6.2# make all make -C linux all ...
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546 views

Asterisk trunk SIP

I have a problem with trunk SIP when I connect my Asterisk to my provider and the Internet connection is lost, the trunk SIP will be unreachable, the problem consist is all local extension are ...
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1k views

asterisk sip port on ubuntu server

I am working on Ubuntu server 12.04 I have asterisk working. I set the port 5060 as this: iptables -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT iptables -A INPUT -p udp ...
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177 views

SIP (G.711) packets getting out of order

I have 2 asterisk servers on 2 different datacenters. Call is going from one to another. Suddenly audio quality decreased. I have made SIP traces on both servers. Here's what I've found: Server A ...
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334 views

Routing Asterisk traffic evading a VPN

I have two offices set in two distant locations, connected through a IPSec VPN (using pfSense). On one office, I have an Asterisk PBX where all the extensions connect, and the Asterisk PBX is ...
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326 views

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 ...
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2k views

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 ...
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3k views

What is the “Allow Anonymous Inbound SIP Calls” option under “Asterisk SIP Settings” in FreePBX for?

I am looking for the canonical definition of the “Allow Anonymous Inbound SIP Calls” option under “Asterisk SIP Settings” in FreePBX. Also, how does it relate to "Allow SIP Guests"?
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255 views

When to limit concurrent calls on Asterisk Server?

I'm interested in building an android app that calls another android app via the Asterisk server. It will be strictly SIP to SIP without connection to a PSTN. I've read you can have thousands of ...
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279 views

Asterisk server warning “Unexpected control subclass”

I am not sure why, but our asterisk server keeps having loads of agents (15) on status ready and calls waiting for them, it seems to be griving us this error in our event log. WARNING[7159] file.c: ...
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418 views

Asterisk can't make nor receive calls through a T1 PRI interface to a Cisco 2430 router

I have an Asterisk 1.8 phone switch with a Digium T1 card. It runs using 5ESS PRI through our present phone provider without a problem. However, we're contemplating switching to Time Warner's fiber ...
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4k views

No voice while making external calls - NAT configuration - Asterisk 1.8.13

As we were trying to setup our Asterisk server, we went on a huge problem: The inability to make call to or from external devices connected to the server. In fact, I can dial and answer the call on ...
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258 views

Asterisk / Voip phone redirects to last called number

I experience a very strange problem with our PBX (Asterisk 1.8, FreePBX, Grandstream GXP1200 phones) If I call an internal number (1020 in this case) the asterisk server redirects me to the last ...
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173 views

Cheap PSTN Card with caller Id for Asterisk Linux server [closed]

For my house I just wanted to setup a caller Id system that would auto-reply on certain callers to avoid harements. Any sugestion for this? I'm thinking on a Linux + Asterisk + PSTN Card with caller ...
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36 views

ASA 5505 base license and Voice vlan [duplicate]

My current setup contains a cisco asa 5505 with two cisco sf300-24p switches, the data and voice vlans have been setup on the switches and I've started to setup the needed vlans on the ASA, but it ...
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5answers
226 views

VOIP Power outage strategy

In the past 2 years, we have had 4 instances where we have been without power for a period of 4-6 hours - due to construction crews cutting power lines, car accidents involving downed power lines, ...
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2answers
102 views

Trans-atlantic VOIP

We're a Canadian business on the east coast and will soon be opening up a new call centre in Australia. This new call centre will be handling our graveyard shift and will be overlapping with the ...
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1answer
144 views

Asterisk 911 dialing rule

Our company is looking to replace our outdated PBX with VoIP. One function we really like about our current PBX is that it alerts the secretary when a line dials 911 and it shows the extension where ...
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340 views

AsteriskNOW server routes outgoing calls as if they are incoming

I am very new to Asterisk so this is undoubtedly a misconfiguration on my end. Right now my test setup behaves as follows: When I make an incoming call to an FXO port on my analog card, Asterisk ...
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226 views

How can I find a log of calls message a specific extension in Asterisk?

Question says it all. Trying to view a log of calls to/from an extension using Asterisk. Thanks in advance.
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1answer
139 views

Polycom IP 650 Custom display

I've been googling all over the place and haven't found a good way to do this. I'm hoping that someone out there has done something like this. I have Polycom 650's and 331 deployed to my users via a ...
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133 views

Variable to store Asterisk extension dialed by user

I am using Asterisk for voice calling. I am using below context If user call back- [from-pstn] exten => _X.,1,Playback(demo-thanks) suppose my number is 74900 on which user can call back. ...
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334 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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1answer
364 views

Why does asterisk (and other systems) feel the need to regenerate in-call DTMF?

I am using Asterisk to interact with analog telephony devices that can be programmed and tested with DTMF interaction. Some of these guys speak rather quickly. Too quickly, you could convincingly ...
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214 views

Troubleshooting a failing SIP connection

I have a linux Asterisk server that is behind a firewall. The firewall has three interfaces : eth0 is the LAN, where Asterisk resides, eth2 is the default gateway to the internet (via an ADSL modem ...
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429 views

Networking issue with asterisk server

I replaced an IVR machine for incoming call after going down. It is running asterisk 1.4.23 on ubunutu 10.04 I decided to put the server behind iptables because my server was under brute force attack. ...
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779 views

Not able to register via SIPclient on asterisk server?

I'm unable to register a SIP client on my Asterisk server. What changes do I need to make to my Asterisk server configuration to fix this? Here are my configuration files for Asterisk. sip.conf ...
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2answers
963 views

PBXact Behind ASA 5505 nat udp range 10000-20000

I'm installing a new pbxact phone server onsite and I've ran into an issue using a 3rd party sip, I've got the acl's for the traffic created and the traffic is allowed, I'm assuming i need to create a ...
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1answer
593 views

Asterisk IVR - input variable and go to an outbound URL

I have an Asterisk outbound IVR system set up, and I'd like it to talk to my LAMP stack on a different server. I've set up the PHP end to receive data from the IVR server and record it in MySQL, but I ...
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1answer
465 views

Asterisk Connection not working

I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something ...
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2answers
6k views

Inbound SIP calls through Cisco 881 NAT hang up after a few seconds

I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's ...
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869 views

asterisk peer with SIP provider through proxy

I'm trying to make a asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but it doesn't seem to cover the case ...
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981 views

Accept calls on external cell phones - FreePBX Distro

We installed FreePBX distro last week and configured it so that when someone calls in, it goes to all the phones so someone here can answer the call. After trying all the internal extensions, it ...
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1answer
221 views

can't register a soft phone to asterisk11

I have a VM (on oracle vbox) running Fedora17. I've installed asterisk 11 on it from sources. I've followed the wiki for installation (https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts) ...