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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, ...


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.


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

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

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 ...


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

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

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

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

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

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


1

38 503 NETWORK_OUT_OF_ORDER network out of order [Q.850] This cause indicates that the network is not functioning correctly and that the condition is likely to last a relatively long period of time e.g. immediately re-attempting the call is not likely to be successful. My guess is that your Caller ID is offending them. Are you setting to anything ...


1

I guess it all depends - I am assuming you are running Asterisk on some sort of Linux? I have done this on several occasions and it can be a bit tricky here and there. But - when it comes to debugging, there are a number of tools that can help. Unknown clients can be blocked using IPTables for example. If you know where your calls are supposed to come ...


1

Well, i found out i was allowing SIP guests. Seems like many bots are looking for unsecured servers to spam or relay. So i just disabled this option, and my logs are now back to a normal state.


1

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)


1

You need allow busy message in extensions tab AND press APPLY CHANGES button on top


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

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


1

In freePBX, these are the steps: Add a system recording (Admin -> System Recordings) Create an Announcement using that recording. (Application -> Announcements) Set the "destination after ring" to whatever you want to (probably whatever extension somebody is picking up) Set the announcement as the destination on the appropriate inbound route ...


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

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

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

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

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

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


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

I use trixbox, and don't use LDAP as of yet it, but the package is called php-ldap. "yum install php-ldap" It may be not quite the same directory for config /var/www/html/web-meetme/lib/adLDAP.php Interested to know the comparison, apologies if it's not a direct answer!



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