Hi again!
On 03.03.2014 18:54, Joshua Colp wrote:
I I know call the phone via PJSIP, asterisk does not reuse the TLS
connection. It tries to create a new one, which of course fails.
What's the exact configuration in use? Do you have a transport
explicitly specified for the endpoint? Doing so will currently cause it
to try to create a new connection [1].
[1] https://issues.asterisk.org/jira/browse/ASTERISK-22658
Thanks for your hint. I tried it, of course:-) But asterisk still tries to
create a new, additional connection to the endpoint. I verify this with a
tcpdump session.
Endpoint/AOR config:
[12300001_3]
type=aor
max_contacts=1
[12300001_3]
type=endpoint
transport=
context=test
disallow=all
allow=alaw
allow=gsm
media_encryption=sdes
rtp_symmetric=true
force_rport=true
ice_support=true
direct_media=false
send_pai=true
auth=12300001_3
aors=12300001_3
After reloading I see with pjsip show endpoint:
...
tos_video : 0
transport :
...
Seems to be correct ;-)
I have 3 transports configured, 2 udp, 1 tls. I also saw this while debugging:
<--- Received SIP request (1112 bytes) from TLS:192.168.203.81:44160 --->
.....
Contact: "A. Valentin"
<sip:[email protected]:58251;transport=TLS;ob>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-00
Look at the port, both ip's are in the same subnet. But this shouldn't make a
difference since I use force_rport=true, or not ?
Do you have any idea ? I'd love to switch to pjsip and gain more experience :-)
Kind regards,
André
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