asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	ce423d2ea4	func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan to use the CHANNEL function on a chan_pjsip channel to obtain run-time information about the channel from the PJSIP channel driver and the PJSIP stack. This includes: * RTP information, including source/destination media addresses, whether or not the media is secure, held, and other properties. * RTCP information. This includes sets of parseable information, as well as individual statistic attriutes. * PJSIP information. This includes URIs, local/remote signalling addresses, whether or not the signalling is secure, and other properties. * The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT function to obtain more detailed endpoint information. Review: https://reviewboard.asterisk.org/r/3038/ ........ Merged revisions 403618 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	2364626811	res_pjsip_t38: Don't pass T.38 control frames through to other hooks. This crept up during gateway testing where the gateway would receive the request to negotiate and assume it came from the remote side, causing the gateway state machine to go a little, to a use a technical term, "wonky". ........ Merged revisions 403364 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	88c840db50	res_pjsip_t38: Add the framehook to the channel only on first INVITE. The check for determining whether the T.38 framehook should be added to the channel or not has now been changed to guarantee adding only occurs on the first incoming or outgoing INVITE. ........ Merged revisions 403258 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	c977f73e13	Fix crashes in res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and external_media_address is set. The callback function for changing the media address in streams wrongly assumes that a connection line will always be present. This is false as no line is present if a stream has been rejected. (closes issue ASTERISK-22645) Reported by: Rusty Newton ........ Merged revisions 400360 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	51a9b93a14	Fix a crash in res_pjsip_t38 caused by the wrong assumption that a session will always have a channel. When starting up or shutting down this assumption is false. ........ Merged revisions 400284 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	391f0003c4	Change the "external_media_address" PJSIP endpoint option to "media_address". The endpoint option does not apply to communication with external entities. Rather, the option is applied to all communications with the endpoint. The external_media_address transport configuration option may override the endpoint option if it turns out that we are going to be communicating with an external entity. Two things of note: 1) I have not updated the XML documentation. This is being taken care of by Rusty as part of his work on issue ASTERISK-22405 2) This commit is likely to cause testsuite failures since there are tests that use the external_media_address endpoint option, and they will need to be changed over. Well, I'm planning to get that updated ASAP after this commit. (closes issue ASTERISK-22528) reported by Rusty Newton ........ Merged revisions 399283 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	735b30ad71	The large GULP->PJSIP renaming effort. The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_" are now "chan_pjsip_" * All files that were "res_sip" are now "res_pjsip" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_" are now "pjsip_" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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57 Commits (3077ad0c2402136f44462c3030fb688d0006e830)