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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

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

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

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

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 ...


1

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


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

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


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

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

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

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

We usually do this by adding contexts in the extensions_custom.conf file. These custom contexts include the default contexts but listen for your outbound calls such as NXXNXXXXXX, 1NXXNXXXXXX and add your dialing codes such as 7777 to the beginning of the call. Then you simply setup your outbound routes so the specific routes are listening for the relevant ...


1

Another thing you can do is change the caller ID information depending on which extension is being called. You can do something like this: exten => 2001,n,Set(CALLERID(name)=Call for Ext. 2001) exten => 2003,n,Set(CALLERID(name)=Call for Ext. 2003) and the incoming number will still show-up, but the name field will show which extension is being dialed... ...


1

Yep that will work just fine. Your dial statement would look like (assuming your peer is named the same as your extensions): exten => 2001,1,Dial(SIP/2001) then for 2003: exten => 2003,1,Dial(SIP/2001) When you get the new phones in you will just have to change the Dial statement on the extension and you'll be good.


1

I don't know have you this already resolved, but i may suggest to check your devices mailbox option on the FreePBX GUI and must be like this "device-number@device". Whenever agent logged in with their ampuser extension the system made a link from directory device-number@device to ampuser@default voicemail directory and link the messages from that device to ...


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!


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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