On 10.01.21 at 22:18 Michael Maier wrote:
Aha, I remembered at the PJSIP level a feature[1] got added when the
Google Voice patch was done. It hasn't been exposed to be explicitly
configurable in Asterisk though.
[1]
https://www.pjsip.org/pjsip/docs/html/structpjsip__tpselector.htm#a1622f416d48eb173aed22b750aa28dfa
In fact the flow functionality[1] may get you closer to how you need
things. There was a bug though with it, so either 18 branch needs to be
used or this change[2] included.
[1]
https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L161
[2] https://gerrit.asterisk.org/c/asterisk/+/15256
The patch was vital!
Great! I'm testing it! I'll be back.
Yeah! That's cool.
myfw*CLI> pjsip show transports
Transport: <TransportId........> <Type> <cos> <tos>
<BindAddress....................>
==========================================================================================
Transport: 0.0.0.0-tls tls 3 184 0.0.0.0:5061
Transport: 0.0.0.0-udp udp 3 184 0.0.0.0:5060
Transport: tflow-001 udp 3 184 0.0.0.0:0
Transport: tflow-002 udp 3 184 0.0.0.0:0
Transport: tflow-003 udp 3 184 0.0.0.0:0
Besides the point that the type is not shown correctly - maybe use * or something else neutral, it's working as expected: no more useless listener ports and each trunk / number for
ISP gets its own port and therefore connection. That was the goal.
But: as long as there where those other normal tls transports, the port in the VIA and CONTACT has been always 5062 - even if the flow transports have been used. That's odd. Now it's
5061 (after those additional tls transports have been deleted) - not sure, where the 5061 is actually coming from :-). But most probably not from the correct source :-) - I think
it's derived from 0.0.0.0-tls.
Thanks
Michael
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