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?
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.
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.
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.”
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.
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.
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.