asterisk

Commit Graph

Author	SHA1	Message	Date
Rodrigo Ramírez Norambuena	eec010829a	AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro. Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723	2015-05-13 16:34:23 -05:00
Mark Michelson	034798e37e	Use SIPS URIs in Contact headers when appropriate. RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific scenarios when we are required to use SIPS URIs in Contact headers. Asterisk's non-compliance with this could actually cause calls to get dropped when communicating with clients that are strict about checking the Contact header. Both of the SIP stacks in Asterisk suffered from this issue. This changeset corrects the behavior in res_pjsip/chan_pjsip.c Review: https://reviewboard.asterisk.org/r/4345 ........ Merged revisions 431426 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	2015-01-29 21:02:23 +00:00

Author

SHA1

Message

Date

Rodrigo Ramírez Norambuena

eec010829a

AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro.

Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723

2015-05-13 16:34:23 -05:00

Mark Michelson

034798e37e

Use SIPS URIs in Contact headers when appropriate.

RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific
scenarios when we are required to use SIPS URIs in Contact
headers. Asterisk's non-compliance with this could actually
cause calls to get dropped when communicating with clients
that are strict about checking the Contact header.

Both of the SIP stacks in Asterisk suffered from this issue.
This changeset corrects the behavior in res_pjsip/chan_pjsip.c

Review: https://reviewboard.asterisk.org/r/4345
........

Merged revisions 431426 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431427 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2015-01-29 21:02:23 +00:00

2 Commits