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How to add a MGCP extension via FreePBX WebGUI


(Benzhu) #1

Environment:
Asterisk: 11.16.0
FreePBX:
FreePBX 12.0.76.4 ‘VoIP Server’

I install AsteriskNOW-612-current-64.iso via VM tool, after install successfully, i can’t find a way to add MGCP extension on Freepbx WebGUI.

i found there is no mgcp.conf under directory /etc/asterisk/
[root@localhost ~]# ls /etc/asterisk/m
manager_additional.conf manager.conf.bak meetme.conf modem.conf musiconhold.conf
manager.conf manager_custom.conf meetme_general_additional.conf modules.conf musiconhold_custom.conf
manager.conf.12.0.1.bak meetme_additional.conf meetme_general_custom.conf musiconhold_additional.conf

but the MGCP driver has installed successfully.

localhost*CLI> core show channeltypes 
Type             Description                              Devicestate  Indications  Transfer    
-----------      -----------                              -----------  -----------  ----------- 
Phone            Standard Linux Telephony API Driver      no           yes          no          
ConfBridgeRec    Conference Bridge Recording Channel      no           no           no          
DAHDI            DAHDI Telephony Driver w/PRI & SS7 & MFC yes          yes          no          
MulticastRTP     Multicast RTP Paging Channel Driver      no           no           no          
SIP              Session Initiation Protocol (SIP)        yes          yes          yes         
Agent            Call Agent Proxy Channel                 yes          yes          no          
Bridge           Bridge Interaction Channel               no           no           no          
IAX2             Inter Asterisk eXchange Driver (Ver 2)   yes          yes          yes         
MGCP             Media Gateway Control Protocol (MGCP)    yes          yes          no          
Local            Local Proxy Channel Driver               yes          yes          no          
----------
10 channel drivers registered.

appreciate in advance, who can tell me how to add a MGCP extension?

thanks
BR
Ben


(Lorne Gaetz) #2

There is no GUI support for that protocol. You would have to configure it manually, then add extensions of type “Custom” and add the custom dial string you require to dial the MGCP extension.


(Dave Burgess) #3

… unless maybe he wanted to write one himself and donate it back to the project. I know how much fun that is, and everyone should try it once. :slight_smile:


(Lorne Gaetz) #4

(Benzhu) #5

ok, thanks for your information. i will try


(Richard Bruce) #6

Just wondering if there has been any change to this and maybe a pointer to creating manual configs.

Thanks,


(Dave Burgess) #7

Not that we’ve heard about. @jfinstrom has posted a lot of good information about creating new modules, but MGCP is such a specific and esoteric connection method that it isn’t likely to happen unless someone with the interest actually does it.


(Malcolm Davenport) #8

To pile-on…chan_mgcp in Asterisk hasn’t been touched (beyond an update to make it even compile) in about 4 years, and it wasn’t something that was in widespread use or good standing, even then. So, that’s one way of saying “don’t be surprised if chan_mgcp fails miserably for interacting with your endpoint.”


(Richard Bruce) #9

Thank you for the replies. It would be nice not to have to replace all of the phones if that day comes but it sounds like, at my level of expertise, that I would be wasting a lot of time.


(Tom Ray) #10

This is basically like the Beta vs VHS or HDDVD vs Blu Ray. Back in the earlier days of VoIP tech SIP, MGCP and H.323 where are contenders for being the go to VoIP option. SIP won that battle hands down and with ease. The other two required a lot of work and needed parts (servers/services) to make work while SIP didn’t.

No one really supports those other two protocols unless you’re in a part of the world where these things were used more vs SIP. But I’d look at what is going to happen when you need to replace these phones, which could be sooner vs later, because you’re going to have a different experience in regards to function, features and supporting it.

So I’d get a jump on that versus waiting until your are “Oh Shit Level 1” because things blew up.


(Dave Burgess) #11

I concur. The cost of building the basic server to match your current deployment is miniscule, so being prepared for the eventuality is probably a good plan. It also gives you the opportunity to look for new phones that provide the services your users are looking for without a lot of training required.

The money isn’t part that always goes bad with these types of upgrades. You can usually get it done for less the one days worth of time and aggravation for your CTO and a couple of technicians. Choose a good phone that looks like your old ones and set it up as much as possible like the old phones. Of course, you can also add simplification rules, like not having to dial ‘9’ for an outside number, or four digit “shortcuts” that can make common local calls a breeze. You can also build in all of the widgets that your current phones can’t do (like built-into-the-system actual conferencing) and other improvements to make your phone clients’ lives better.


(Richard Bruce) #12

Thank you for the replies and great advice. I should be considering the features, compatibility and serviceability.