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| author | Sebastian Damm <damm@sipgate.de> | 2016-05-10 17:19:48 +0200 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2016-05-12 05:34:24 -0500 |
| commit | a94a12bbf73c7e643a77e4aa0597e4e49eb5b7ab (patch) | |
| tree | 9b13337d9b3db29a3603d9d0c8919f58f62b096c /.cleancount | |
| parent | a01ce2b88912cd802bb045e40fe264906e55bc45 (diff) | |
res_pjsip_outbound_registration: generate correct Contact URI for TLS
There are two types of SIP URIs indicating a secure transport:
* sips:user@example.org
* sip:user@example.org;transport=tls
When using a sips URI, Asterisk checks incoming INVITEs and answers from
the other side for sips URIs, and rejects the packet if there are only
sip URIs. So Asterisk should only generate a sips Contact URI if the
other side supports it.
This patch makes Asterisk generate either a sip or sips Contact URI
depending on the format of the server URI.
If you want a sip URI, use:
server_uri=sip:example.org\;transport=tls
If you want a sips URI, use:
server_uri=sips:example.org
ASTERISK-25990 #close
Reported-by: Sebastian Damm
Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2
Diffstat (limited to '.cleancount')
0 files changed, 0 insertions, 0 deletions