You might be right if we are only taking within US. However, if you have to have a conference with some participants in the UK some in US and maybe elsewhere it's hard to beat what asterisk offers. But yeah, probably not on RPi
So then, if I understand correctly in the XIVO build ConfBridge is there for "2663", but if you try to use the Xivo GUI to create Conference Rooms 2664 or 2665 etc., it fails, and that is because Xivo tries to use Meet-me which does not work on Raspberry Pi's?There is NO meetme hardware support on a Raspberry Pi so the only available option is a ConfBridge.
That's correct. You can clone the 2663 line in /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf to add additional conference bridges. Or you can use the free Anonochat conferencing service which is much more reliable for conferences than what you'll get with a Raspberry Pi.So then, if I understand correctly in the XIVO build ConfBridge is there for "2663", but if you try to use the Xivo GUI to create Conference Rooms 2664 or 2665 etc., it fails, and that is because Xivo tries to use Meet-me which does not work on Raspberry Pi's?
;# // BEGIN Conf 2663 exten=2663,1,Answer same=n,Set(CONFBRIDGE(user,music_on_hold_when_empty)=yes) same=n,Set(CONFBRIDGE(user,music_on_hold_class)=default) same=n,ConfBridge(2663) same=n,Hangup
You should be just find with 3-4 simultaneous calls. Limit is 10 for high quality on a Pi 1. Probably 20-30 calls for a Pi3.What kind of call volume do you think a RasPi3 could reliably support? I have a two tenant small office with about 15 extensions. Prob no more than 3 or 4 calls at a time. SIP trunks and POTS (will need to find a hardware solution... usb?). Would a RasPi3 be a good choice, or no?
What is the performance of Asterisk running on the Raspberry Pi?
In a typical setup with RasPBX, 10 concurrent calls are possible on a Pi 1. This is also the case for conferences, meaning 10 participants can join a conference. More than 10 calls do work, but audio quality decreases considerably with every additional call. See also:
Source: http://www.raspberry-asterisk.org/faq/How do I interface the RPi with an analog line from my telecoms provider?
Up to date there is no hardware available that is interfacing with an analog line and can be directly connected to the RPi. Calls have to go over Ethernet using any of the VoIP protocols supported by Asterisk. Devices with a PSTN FXO port translating the analog line to SIP are for example the Linksys SPA3102 or the Obihai OBi110. These can be configured as SIP trunks in Asterisk.
Not much need any more. The Wazo releases all contain the Opus codec which is far superior.I know this thread is a little old... but,
I guess there's no chance of compiling g729 for XiVO?