I believe asterisk is evolving way from a "carrier" mindset more into an
IT one. Losing support for ancient hardware is part of that. I no longer
use IAX (used it to trunk multiple instances across vpn's - worked
well.) Currently I only have a couple of now quite old Cisco phones and
Jami softphones on android using SIP to a single asterisk.
I incrementally upgrade asterisk mostly by a clean install carrying over
config files into an LXC instance (using a golden master setup) which
has been trouble free until pjsip - and thats more my fault in missing
that SIP was deprecated and having to unexpectedly fault find it. The
conversion script worked fine for internal extensions, but the uplink
trunk required a few hours extra work until I stumbled over the correct
syntax - too many options confusing things.
My recommendation would be to spin up a VM and install a test instance
and start afresh - older asterisk versions are usually a security risk
as time goes on.
BillK
On 4/2/24 06:00, Thelma wrote:
I think I was able solve my problem, it was as simple as disabling
"jitterbuffer" in iax.conf
I can hear phone voicemail request from the remote asterisk, will know
100% on Monday.
Regarding Asterisk I'm on 16.30.1 ; tried emerging ver.18 but my
AudioCodes box wouldn't even register to it.
Conversion scrip (sip->pjsip) will not do any good if the hardware
(AudioCodes boxes, Sipura and other units) are not compatible with pjsip.
Is IAX is gone as well in newer versions?
I have an impression this is the end of old Asterisk that Digium
started; very, very sad :-/
It had good community support.
Newer versions 18+ are not compatible with older hardware and learning
curve/conversion is not worth it.
Sangoma - community support is almost not existent, few folks just
bark at you if one mention still running ver. 16
I'll hang on to 16.30.1 as long as I can.
On 2/2/24 20:55, William Kenworthy wrote:
In v18 sip is still present but deprecated - after this its removed.
There is a conversion script (sip->pjsip) for migration. It required
a few sacrificial chickens and much swearing until I got the upstream
trunk to register (iinet in AU). Its all working good now, the pjsip
config is more programmer friendly but also allows much more complex
(read hard to follow/fault find) configuration.
Note that the CLI commands are not equivalent to sip (including help,
its now pjsip) with a different format. After install, but before
re-configuration everything sip related disappears on restart.
BillK
On 3/2/24 09:42, John Covici wrote:
On Fri, 02 Feb 2024 19:29:24 -0500,
Thelma wrote:
When did they implement switch-over from sip to pjsip?
I'm using AudioCode boxes.
I emerged and tried to load asterisk ver.18 but the audiocode
would not register. I suppose ver.16 is the end of the line for me.
On 2/2/24 16:39, William Kenworthy wrote:
Yes, was caught out recently by the replacement of sip with pjsip
- currently on v21.0.2 and working (sip only, simple home setup)
Also had some weird problems with two versions installed (so
asterisk started on old working version even though new one was
installed - once I ran depclean it failed due to the sip/pjsip issue.
BillK
On 2/2/24 23:26, Thelma wrote:
Anybody on the list using Asterisk?
I need some help.
Have save version of asterisk is working correctly on one
computer but the other.
I would use at least asterisk 18 in all cases and if you can later
versions. pjsip has been the preferred version for a while, sip is
still OK, however.
