Hello, We would like to install Asterisk 11+ on the QUAD ARM Udoo card but we can not install DADHI, a component that is used for conversations and to create "meetme" rooms. The operating system installed is UBUNTU 12.04 with kernel version 3.0.35. We also tried version Ubuntu 14.04 but there are same errors in the installation phase.
Conference bridges no longer require DAHDI in newer Asterisk releases -- the old "MeetMe" application did, but that's been deprecated for a while, replaced by "ConfBridge" which operates entirely in use-mode without a DAHDI timing source. Its invokation syntax is similar. See https://www.voip-info.org/asterisk-cmd-confbridge/ An example of a simple conference extension: Code: exten => 680,1,Answer exten => 680,n,Wait(0.7) exten => 680,n,ConfBridge(680,sc) The only thing you really need DAHDI for now is if any TDM telephony ports (T1, E1, analog FXS or FXO) are directly attached to the Asterisk server (rather than behind an ATA, SIP gateway etc. over the network), using something like a Sangoma or Digium board. This isn't really applicable to the Udoo, since it lacks a PCI bus (its i.MX6 SoC includes a single PCI-e lane, but it isn't bonded out), though there are a few USB FXO/FXS devices that might work, and would require DAHDI if so. I've toyed with the idea of building some sort of telephony shield to provide direct FXS/FXO ports, perhaps using "SLIC" chips from Silicon Labs, something analogous to this offering for the Raspberry Pi (which connects via SPI for control, and I2S for audio streaming - would Udoo's SSI and AUDMUX channels work?)-- https://switchpi.com/product/oak8x-base-module-4-fxs-module-plugged-but-without-onboard-4fxo/ but it's hard to justify when I have a pile of old Linksys PAP2 ATA's lying around that work well enough. The only real benefit would be a slightly reduced power consumption, freeing one or more Ethernet switch ports, and a tiny reduction in end-to-end latency. (Direct SIP phones, such as Grandstream's multi-handset SIP DECT offering are superior in many ways, especially in an office setting, but aren't always ideal for residential use. My husband wouldn't like having to hit transfer / conference / barge-in softkeys rather than just picking up handsets at will to join a call.
I'd be interested to hear your experiences with Asterisk 11, though - upgrading all or some of my Asterisk sites (currently still on 1.8.xx) to that version is on my long-term to-do list. I'm mainly interested in the new support for SMS over SIP (maybe IAX2 as well? The only Asterisk server I run with DAHDI is the one that has a T1 PCI card in it, and that's soon to be disconnected anyway in favor of SIP trunking.
