I wouldn't mind using SIP with voip.ms if it worked right. I'll try it
again changing what needs changing, but for now, I'm not conducting any
secure communications, so I'll worry about it when things break.
On 5/27/2023 11:20 AM, aster...@phreaknet.org wrote:
IAX2 tends to work really well for trunking. Unlike SIP, it usually
just works, although it tends to be more a niche use case. For this
reason, IAX2 has long been a controversial technology; most people
seem to either love it or hate it. Obviously, you can guess what my
The only downside in your case is voip.ms's IAX2 stack (whether
Asterisk or something else) does not support encryption, and it does
not appear they have plans to support it. If you don't mind that, it
shouldn't be an issue.
voip.ms is also the only major VoIP provider that supports IAX2, so if
you do anything else you'll probably have to use SIP.
On 5/27/2023 10:23 AM, Steve Matzura wrote:
I'll take that under advisement, but Doug swears by IAX, I tried it,
it worked, so until things break and break bad, I'll stick with that
and try the recommended remedy, now recommended by two people.
On 5/26/2023 8:08 PM, Sean Bright wrote:
On 5/26/2023 5:41 PM, Steve Matzura wrote:
Doug from this list got me to change my connectivity to my DID
from SIP to IAX, and bingo, it all just worked instantly.
Looking over your previous messages and the error you were receiving
(the one referring to extension 's') it looks like you had your
VoIP.ms account setting incorrectly configured. There is a "Device
type" dropdown that needs to be set to "IP PBX Server, Asterisk, or
Softswitch." If instead it is set to "ATA device, IP Phone or
Softphone" (the default) then it will be sent to the 's' extension
instead of the DID one. I captured a screenshot¹ from my account.
I created a VoIP.ms account, acquired a DID, copy/pasted the VoIP.ms
configuration samples², substituted my SIP Account User ID and
passwords, restarted Asterisk, and everything worked as expected.
I would never recommend new installs use IAX2, so if you envision this
moving beyond the toy/PoC stage I suggest you giving PJSIP another go.
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
asterisk-users mailing list
To UNSUBSCRIBE or update options visit: