Hot answers tagged

21

Yes. Being in a position currently where the powers that be bought a Panasonic PBX. There's a bunch of features that aren't there (or rather, are individually licensed). Such as MeetMe (conference calling), or hot-desking. There's also the problem that all the phones are effectively hard-wired to their ports. They had to run new cables for the phones, ...


8

Don't worry, it'll be iron age performance, at worst. (grin) A large number of netfilter rules, which the kernel has to run through for every single packet, will have a negative impact on system performance and resource consumption (both CPU and memory). The nice thing is that there are a number of tricks which you can employ to minimise the problems. ...


7

Your networking issues will increase exponentially. You'll have to troubleshoot issues with latency from various locations, you'll have worse packet jitter, etc. I've had nothing but heartache with SIP to remote sites/servers.


5

The main issue with iptables rules is that they are executed in sequence and with a large rule-set odds are quite a few rules have to parsed before a packet is either granted access or rejected. Womble's answer already explains quite a few strategies to reduce that processing penalty, by cleverly ordering rules and where I agree the most important one is ...


4

I would separate your question into 2 areas: Going VoIP Going Open Source Regarding the first one, I believe there is not much dispute, being VoIP extremely more powerful than traditional PBX, but to name a few: Usage of same network infrastructure you already have laying around, reducing cost Ability to connect remote endpoints without expensive ...


4

Trixbox is pretty good (http://www.trixbox.org/). It's basically Asterisk but shake'n'bake style. Their community edition is a free version which installs Asterisk plus some fancy front-end tools among other things. It has IVR (auto-attendant), SIP support, remote extensions and other cool stuff. Try it out.


4

Yes, this is called DISA and is built into Asterisk. Here is a walkthrough. If you have FreePBX, you can do this through the GUI...look on the left side under "Internal Options and Configurations."


4

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 ...


3

Here's how I did that. My office has a static IP address, so my home server will be connecting to the office. This should fit most people with dynamic IP addresses at home. I did this from the office and created an SSH tunnel to the web interface of my trixbox home server: ssh -L8080:trixbox:80 me@home. Open UDP port 4569 in the office firewall Open UDP ...


3

is it dependent upon configuration? It surely is. And the configuration will depend on your requirements. If you do recoding, call recording or if your external users are behind a NAT or unable to contact your SIP provider directly, you can't bypass the PBX in the media path. The asterisk extension option for a direct media communications path is ...


2

Shameless plug :) but you could try my product SAMReports, it works with FreePBX, as well as with other Asterisk editions .


2

You're limited by the number of analog lines you have connected. Each line can handle only one call at a time - so no matter what settings you change, your system won't have any higher external capacity. Note that if any call has an endpoint outside your organization - whether an inbound or outbound call - it needs a free line. You need a bunch of pieces - ...


2

The important parts of your log are these two WARNING messages: [2012-08-01 09:08:11] WARNING[3921] chan_sip.c: Retransmission timeout reached on transmission 3e57de11239b49f10930969c7e23d1ca@50.58.42.81:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 9983ms with no response ...


2

Generally you're best off bringing both machines to the same version before trying to restore. If you don't go that route the thing to watch out for is schema changes between versions. Read the changelog carefully, and mount a scratch monkey before sacrificing your production environment (while virtual machines won't work for VOIP they are fine for ...


2

I'm sure I must did something when I was trying to configure a new DID from Voip.ms. Suddenly my sip phone can't dial out. Tried to call extension to extension and having the same problem also. Yup, sounds like you trashed your configuration while setting up the voip.ms stuff. Undo your changes (get your system back to a working state), then contact ...


2

There are two steps needed to fully remove the "sip:+" prefix... Removing the Plus Prefix FreePBX includes a "context" which strips all but the final 10 digits from the CID string. There are several other threads out there, which say to create a custom context, but this is not necessary. On your inbound route, if you change this... context=from-trunk ...


2

First off, you probably want to switch your system to "Device and User" mode (see here). It is a little more complicated than the default "extensions" mode, but substantially more flexible. In Device and User mode you will have an option on the User page to enable voicemail: (I believe you'll have the same option on the "Extension" page). Set the ...


2

Even with GUEST turned off, your Asterisk server is still going to be hit with SIP hacking attempts. You should install something more to monitor and manage the attacks. Have a look at Asterisk Security for suggestions and even the basics of how to secure your Asterisk server. (Guest off is NOT enough)


2

If you work with FreePBX you will most likely need to go under the hood and add a context like below to /etc/asterisk/extensions_custom.conf [check-ending] exten => _!,1,GotoIf($[${EXTEN:-6} = 777777]?gotcha) exten => _!,n,Goto(inbound) exten => _!,n,Hangup() exten => _!,n(gotcha),Goto(inbound-from-777777) exten => _!,n,Hangup() exten => ...


1

All the asterisk frontends I have used generated asterisk config files from their own config database. They don't work the other way around. If you want a GUI you are likely stuck reconfiguring asterisk from scratch using the new GUI and manually setting all the settings your existing config has.


1

You should switch to queues instead of ring-groups for this task, you can confirm calls in queues just make the first queue failover to the second and make the cell phones call in an order. 5555551111,1 5555551112,2 More on queues module http://wiki.freepbx.org/display/FPG/Queues+Module+User+Guide


1

Of course, you will not only need the second card but 4 more working POTS telephone lines. If you might want more lines than four times the number of suitable slots in the server, consider renting ISDN lines instead if available, there are cards available which are quad BRI so eight voice channels to a card (and even quad PRI cards which do a ludicrous ...


1

If by "calling" you mean from the PSTN, then you will have to add another card. It should be pretty much as easy as adding the card and using the manual to tell you how to get Asterisk to use it.


1

You need just strip digits. i.e in freepbx do rule 1933+NXXXXXX in the TRUNK


1

To generate a call using the Asterisk Manager Interface (AMI), first you will need to write a dialplan context in /etc/asterisk/extensions.conf that performs the desired actions after the call is connected. (Playback a sound, then hangup.) From the external system, use AMI to Login and Originate a call to the desired destination. In the originate action, ...


1

If you want the 933 and 1 added the you'd need "Outbound Routes" like this: NXXXXXX NXXNXXXXXX Then, in your Trunks, you want this in the diaplan rules. 933+NXXXXXX 1+NXXNXXXXXX That would accept NXXXXXX as a valid number from your handsets but would dial 933NXXXXXX when the call goes out on the trunk. The same goes for the 1. It will be added to all ...


1

Alright, a different approach: Create an announcement. Add extension (virtual) Give it a number, make it so its no answer destination is your announcement. Go to trunks Add custom Destination: "Local/{ext}" (replace {ext} with extension number, ex: Local/1000) Add an outbound route, see dial patterns below, first route (so it catches all calls), ...


1

I'd recommend just getting the new FreePBX distro. If you are set on using FreePBX, it's the best, most stable, best supported way to go. Otherwise, there are tutorials on http://freepbx.org for installing it from source on debian: http://www.freepbx.org/support/documentation/installation/install-process-for-debian There shouldn't be any reason why you ...


1

Freeswitch is probably a better fit, and isn't based on Asterisk. The biggest problem will probably be that the majority of VPS providers are utter shite.. Especially when things like uptime, and network availability are concerned. There are other problems too, such as IO and network contention, which could mean that you'd drop calls or call quality, ...


1

Yes you are correct. Once you define 0.0.0.0 as the listener, then trixbox will listen on both interface IP addresses for incoming connections. This is the default behaviour. It is actually the opposite option that you would need to change the config. If you had two interfaces and only wanted to listen on one, then you would update bindaddr to have the ...



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