Asterisk 11 available

[Posted November 2, 2012 by n8willis]

Digium has unveiled version 11 of the free software telephony server Asterisk. This is a Long Term Support (LTS) release, which means support will be provided for four years. Changes in this version include WebRTC support, the DTLS-SRTP secure transport, and channel drivers for Jingle and Google Talk.

Asterisk 11 available

Posted Nov 2, 2012 22:46 UTC (Fri) by rillian (subscriber, #11344) [Link] (2 responses)

Pity they dropped Opus support.

Asterisk 11 available

Posted Nov 3, 2012 0:03 UTC (Sat) by pabs (subscriber, #43278) [Link] (1 responses)

Do you have any references about that? Was the support not ready yet? What was the reason for dropping it?

Asterisk 11 available

Posted Nov 3, 2012 2:55 UTC (Sat) by rillian (subscriber, #11344) [Link]

Not really. dvossel started on support about a year ago, but then removed it from his branch in the repository. I asked about this on irc, and was told that there were concerns about IPR, and Digium would have to do its own review of the issue before support could be considered.

It appeared to me the maintainers either fell victim to FUD, or read the IETF declarations page for the draft without appreciating the niceties of that process as described on opuscodec.org. I've tried to mirror the relevant portion of svn history here if anyone is curious.

This was all months ago; it may be that they're ready to consider support now that the new release is out, but it's unfortunate not to have this exciting new codec in the long term support release. Nevertheless we hope to demonstrate Opus streaming through Asterisk at the IETF meeting next week, so there is working code outside the official repo.

Asterisk 11 available

Posted Nov 3, 2012 0:17 UTC (Sat) by imitev (guest, #60045) [Link] (4 responses)

last time I checked they were at asterisk 1.8 ; looks like that they're now following mozilla/google chrome naming spree.

off-topic - having a project beginning with "A" is a double-edged sword: you're at the top of software lists, but you're also at the top of LWN vulnerabilities list. I can't recall the number of times I've seen asterisk listed when I was reading the articles on the security page.
I know asterisk is a big project, but making it so often on the list is quite frightening. That, plus their NIH syndrom and (at the time of 1.2) bad support even for customers who bought a few thousand dollars of digium hardware (like me), made me switch to freeswitch (!) and since then I've never looked back.

Asterisk 11 available

Posted Nov 3, 2012 0:41 UTC (Sat) by prometheanfire (subscriber, #65683) [Link]

Asterisk being at the top of vuln lists isn't just because of starting with the letter 'A'. They still have the NIH syndrom and don't have a good regression testing policy. Even if a patch is provided to them for an issue it's like pulling teeth to get them to apply it. How I wish I could switch to freeswitch, but their bundled libs really irk me (moreso then Asterisk problems...). Have fun with FS though :D

Freeswitch?

Posted Nov 5, 2012 14:15 UTC (Mon) by dskoll (subscriber, #1630) [Link] (1 responses)

switch to freeswitch (!) and since then I've never looked back.

How painful was that switch? We currently use Asterisk and have a whole bunch of integration tools that work with it, but the annoyances are building up. So what's your experience with switching? And how about daily operations... do you like Freeswitch?

Freeswitch?

Posted Nov 8, 2012 10:50 UTC (Thu) by imitev (guest, #60045) [Link]

hm, forgot to enable mail notifications, sorry for the late reply

FS' wiki states that people knowing asterisk are disadvantaged compared to people coming without any prior background (except of course voip knowledge), and that's quite true: I scratched my head many times trying to understand how FS did things. It doesn't mean that FS is more difficult to configure than asterisk, it's just that people are usually bored re-learning things.

I was a little bit annoyed by the all-XML config style but in the end you'll find out that it's a rather good technical decision as soon as you delve into more complicated setups - and that's where really FS shines. For instance, I had to use a standalone sip proxy because asterisk's sip configuration didn't support multiple external ips (the voip server was NAT'ed in a DMZ, and sip clients could connect through different internet providers, with/without VPN). FS handles that by design and provides a clean separation between sip instances (running each on a different port).

For sure that post doesn't help you much (and since I don't know your voip setup I can't give you specific examples), but yes, I like FS more than asterisk and I didn't have any stability problems. Security seems to be taken into account at the architecture level, and FS is really flexible, so no worry for complicated setups.
WRT integration tools, we had custom scripts/interfaces so we could adapt them to FS (LDAP user management, accounting, ...). If your setup is complicated I'm sure the devs will help.
On the downside, FS doesn't have asterisk's critical mass, so good luck finding an admin who knows FS :(

Asterisk 11 available

Posted Nov 5, 2012 21:49 UTC (Mon) by dfsmith (guest, #20302) [Link]

I think another reason for Asterisk being top of the vulnerabilities list is that it is a big target. There is *real money* for grabs by hacking phone accounts.

When I opened the SIP port in my firewall, I was inundated with password cracking attempts (mostly from IP addresses in Scotland, bizarrely).

I gave up on the well-known SIP port, and went with a non-standard, fail2ban covered port when I got bored listening to my honeypot SIP account receive forwarded calls.