13

Start by having a look at this voip-info web page: high availability design. It will explain what high availability is and isn't - in the context of Asterisk. (Its easy to confuse high availability with load balancing) Next, look at this voip-info web page: high availability products. It will explain what is out there for open source / commercial ...


10

The PBX is going to handle all of your endpoints, your hunt groups, your schedules, your auto-attendant, call flows/trees, recordings, etc.. Your SIP provider gives you phone lines. The PBX allows you to use those phone lines. A lot of SIP providers can also provide hosted PBX services, so they handle everything. You could set up a PBX in your office for ...


6

You forgot to set localnet. The configuration file did warn you that you have to set this equal to the netblock of your local network. ;----------------------------------------- NAT SUPPORT ------------------------ ; ; WARNING: SIP operation behind a NAT is tricky and you really need ; to read and understand well the following section. ; ; When Asterisk is ...


5

The simplest, lazy way around this is to set your asterisk box in a 1:1 nat config (often called DMZ host on home routers) and to make sure externip= is set in sip.conf. The usual troubles with SIP and NAT are: SIP headers contain call source and destination information (IP addresses) that may not be reachable to/from clients and servers behind nat A ...


5

Since I've been answering the question in the comments I figured it was more appropriate to put it here: It's not unusual for your PBX (or any service) to get probed by bots/hackers/whatever out there when you are internet facing... This happens with web servers, SSH, SMTP, etc... all the time. As long as you are open to the Internet without restriction, ...


5

It's a brute force, asterisk servers always get that kinda thing if connected to the public IP. My solutions are : Install fail2ban, fail2ban will set the iptables and reject ip with continous fail attempts to the asterisk Disable guest sip login, put allowguest=no in the sip.conf In case if you're using VPN, set the from sip external in your PBX Set the [...


4

Digium has somewhat complete documentation for all AMI Actions and AMI Events on their wiki. I say "somewhat" because it lists all available actions and events, and how to use them, but some of it is a little sparse when it comes to the output you should expect to see. Actions you might find useful: Command - lets you run an arbitrary CLI command from the ...


4

You ask multiple questions, I can answer some, but not all of them. why asterisk is running on loopback address and not on 192.168.32.181:5038 As far as I can tell from a few searches port 5038 is the asterisk management interface. So I assume it binds to 127.0.0.1:5038 for security reasons. You wouldn't want that to be accessible from everywhere. Why ...


4

Try adding the valuehost=dynamic to your guest config in iax.conf


4

If you just want a simple phone, a VoIP equivalent of just renting a single phone line and plugging a dumb phone into it, you don't need the PBX, just a Sip Trunk. If you want the features that the PBX brings to a traditional phone line on your VoIP line, you need a PBX for that too. Some Sip Trunk provides provide a few of these features at the host end ...


4

You need what Asterisk calls a telephony interface card ; a PCI or PCI Express expansion card that connects the server running Asterisk directly to your (legacy) phone line(s). Depending on your locality you might be able to port you existing landline number to a (current or new) VoIP provider and save yourself the hassle maintaining the existing ...


3

It is because not all legs / providers properly cancel echo in their network. So, in theory you should not need to have a line side echocan on a digital line, but in practice you sometimes do. For example, a provider might not cancel echo out to each analog endpoint in their network. If you have a digital connection to this network, you get the signal ...


3

Try it with quotes, like this: "username":"password"@server


3

There isn't an easy answer to this, but a number of people have suggested solutions. Basically you need to count the outbound channels yourself, as suggested here: http://www.remiphilippe.fr/2010/05/29/simultaneous-call-limitation-on-asterisk/ The script looks like this, after groups have been enabled as a macro: [globals] MAXCALLS=2 [macro-voipcall] ; ...


3

This condition happens when ari or ampm passwords were recently changed and it contains some illegal characters that asterisk would not recognize like semicolons. It is safe to use passwords including letters, numbers and underscore. Just check /etc/asterisk/manager.conf and /var/www/html/recordings/includes/main.conf.php for passwords.


3

Got the answer. My manager.conf now looks like this: [general] enabled=yes port=5038 bindaddr=0.0.0.0 displayconnects=yes ;only effects 1.6+ [admin] secret = abc123 deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 read = system,call,log,verbose,command,agent,user,originate write = system,call,log,verbose,command,agent,user,originate Also, when ...


3

Ookay, I figured it out. This is just workaround, but the problem has dropped to acceptable level for me. After un-wanted reboot in a middle of a day I noticed that Load Average dropped greatly! I checked our server monitoring system - LA always climbed up little-by-little in a 3-4 hours of continuous (and stable) load. This was un-noticable, because I was ...


3

Thank you for all of your help. Finally, I read somewhere that it may be helpful to record calls, so I started using the Monitor() application. Finally, I found that Asterisk wasn't listening for silence until after it played a recording first. I found this by realizing that Monitor() didn't start recording until something was first played by Asterisk, and ...


3

Lack of audio in SIP calls is nearly always due to firewall or NAT issues. All traffic to/from your EC2 instance traverses a NAT, so I'd look into making sure that the requisite NAT setting in Elastix are set properly.


3

If you can hard-wire, you should. Doing so is the only way to get guaranteed bandwidth and latency, both are which impossible to guaranty in the "unlicensed" wifi bands. This is not VoIP specific, but holds true for all aspects of networking.


3

1. Well.. if there are failed auth attempts from that number, but you're getting calls from it "from time to time", there's a chance that sometimes, the auth attempt is successful. Try catching the ip address that registers with it by periodically checking asterisk -rx 'sip show peers' for that number. EDIT (probably a better idea): When the phone rings, ...


3

Right now, you only have 1 extension defined in your [test] context ... Extension 20. If you want to be able to place calls between 200 and 201, you'd need to have something like the following: [test] exten => 20,1, Answer() same => n, Playback(hello-world) same => n, Hangup() exten => 20[01], Verbose(2,Call for Extension ${EXTEN}) same =&...


3

A lot of people think a firewall is a security system for a PBX. It's not. If you forward SIP and RTP into your PBX through your firewall, then your firewall is only acting as a router (from the perspective of VoIP). There is no checking of valid users, devices, geographic locations, dialing patterns, user behaviours, etc. Setting up basic security for ...


3

There are multiple logs in /var/log/asterisk, check them all Debugging from the console You can also start asterisk from the console and enable debugging for iax2 # asterisk -r asterisk*CLI> iax2 set debug on asterisk*CLI> Then try connecting again from your client. You should see more info on the console. For more info in iax2 debugging asterisk*...


3

You absolutely can do this. Whether it's appropriate for your situation is a different matter. Essentially, SIP trunks are like POTS lines or T1/PRI trunks. They are just dial tone. They can be used (and combined) in various ways depending on the business application. Using that analogy: You could run a individual POTS lines to each desktop (i.e. associate ...


3

This happens due to automated scanners, which probably are trying to bruteforce your passwords. To get rid of such calls, you should disable anonymous users by placing the following options into the [general] section of your sip.conf [general] context=bogus allowguest=no alwaysauthreject=yes


3

Tono, ufw is a front end for iptables. What you are asking essentially, is why does DNS fail when you stop accepting Related,Established connections. conntrack keeps track of your connections. If traffic matches an entry in conntrack then it is Established. If an Established connection creates a new connection then it is Related. If the traffic is new and ...


3

First, the register line should have a path set at the end, like: register => myusername:mypassword:myusername@sip.flowroute.com/84106639 Then do a sip reload or service asterisk restart. After that, the sip show peers command should return some kind of status. The qualify=yes option is useful too to check IP connectivity and SIP service status. Also, ...


3

You are missing two items to be placed directly under your register => entry. [general] register => myusername:mypassword@sip.flowroute.com registertimeout=20 registerattempts=0 As soon as you enter these in sip.conf, at the asterisk console enter: *CLI> sip reload *CLI> sip show registry


Only top voted, non community-wiki answers of a minimum length are eligible