From 7ce04c1641d67df68eac94c5bf5f8aff8fd44d43 Mon Sep 17 00:00:00 2001 From: Mark Michelson Date: Tue, 19 Jul 2016 10:29:54 -0500 Subject: [PATCH] ChangeLog: Updated for certified/13.1-cert8 --- ChangeLog | 157 ++++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 157 insertions(+) diff --git a/ChangeLog b/ChangeLog index 4369835855..01a89d9f00 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,160 @@ +2016-07-19 15:29 +0000 Asterisk Development Team + + * asterisk certified/13.1-cert8 Released. + +2016-07-19 10:29 +0000 [6837d582bc] Mark Michelson + + * Release summaries: Remove previous versions + +2016-07-19 10:29 +0000 [81bda18275] Mark Michelson + + * .version: Update for certified/13.1-cert8 + +2016-07-19 10:29 +0000 [8cfe7d9aac] Mark Michelson + + * .lastclean: Update for certified/13.1-cert8 + +2016-07-19 10:29 +0000 [cca43b1a2f] Mark Michelson + + * realtime: Add database scripts for certified/13.1-cert8 + +2016-07-07 10:38 +0000 [690e23433b] Joshua Colp + + * chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled. + + Some T.38 implementations may send another re-invite after the initial + one which adds additional negotiation details (such as the max bitrate). + Currently this will fail when passthrough is being done in chan_sip as we + do nothing if T.38 is already active. + + Other handlers of T.38 inside of Asterisk (such as res_fax) handle this + scenario so this change adds support for it to chan_sip and res_pjsip_t38. + If a request to negotiate is received while T.38 is already enabled a + new re-INVITE is sent and negotiation is done again. + + ASTERISK-26179 #close + + Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c + +2016-06-02 12:51 +0000 [8b16e992c8] Richard Mudgett + + * pjsip_distributor.c: Ignore messages until fully booted. + + We should not be processing any incoming messages until we are fully + booted. We may not have dialplan or other needed configuration loaded + yet. + + ASTERISK-26089 #close + Reported by: Scott Griepentrog + + ASTERISK-26088 + Reported by: Richard Mudgett + + Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264 + +2016-05-18 07:54 +0000 [08a7e1d0b0] gtjoseph + + * udptl: Don't eat sequence numbers until OK is received + + Scenario: + Local fax -> Asterisk w/ firewall -> Provider -> Remote fax + + * Local fax starts rtp call to remote fax + * Remote fax starts t38 call back to local fax. + * Local fax sends t38 no-signal to Asterisk before sending an OK. + * udptl processes the frame and increments the expected sequence number. + * chan_sip drops the frame because the call isn't up so nothing goes out + the external interface to open the port for incoming packets. + * Local fax sends OK and Asterisk sends OK to the remote fax. + * Remote fax sends t38 packets which are dropped by the firewall. + * Local fax re-sends t38 no-signal with the same sequence number. + * udptl drops the frame because it thinks it's a dup. + * Still no outgoing packets to open the firewall. + * t38 negotiation fails. + + The patch drops frames t38 received before udptl sequence processing + when the call hasn't been answered yet. The second no-signal frame + is then seen as new and is relayed out the external interface which + opens the port and allows negotiation to continue. + + ASTERISK-26034 #close + + Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9 + +2016-05-17 11:14 +0000 [ce3620caab] gtjoseph + + * chan_sip: Prevent extra Session-Expires headers from being added + + When chan_sip does a re-INVITE to refresh a session and authentication + is required, the INVITE with the Authorization header containes a + second Session-Expires header without the ";refersher=" parameter. + This is causing some proxies to return a 400. Also, when Asterisk is + the uas and the refresher, it is including the Session-Expires and + Min-SE headers in OPTIONS messages which is not allowed per RFC4028. + + This patch (based on the reporter's) Checks to see if a Session-Expires + header is already in the message before adding another one. It also + checks that the method is INVITE or UPDATE. + + ASTERISK-26030 #close + + Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9 + +2016-05-05 05:07 +0000 [6d4928a135] Joshua Colp + + * file: Ensure nativeformats remains valid for lifetime of use. + + It is possible for the nativeformats of a channel to change + throughout its lifetime. As a result a user of it needs to either + ensure the channel is locked when accessing the formats or keep + a reference to the nativeformats themselves. + + This change fixes the file playback support so it keeps a + reference to the nativeformats when accessing things. + + ASTERISK-25998 #close + + Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915 + +2016-02-15 15:37 +0000 [3294ac1efe] gtjoseph + + * res_pjsip_caller_id: Fix segfault when replacing rpid or pai header + + If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid + or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify + the header added by the dialplan function. Since the header added by the + dialplan function is generic string, there are no virtual functions to parse + the uri and we get a segfault when we try. Since the modify, was really only + an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER + and recreate it. + + This raises a question for another time though: What should happen with + duplicate headers? Right now res_pjsip_header_funcs doesn't check for dups + so if it's session supplement is loaded after res_pjsip_caller_id's (or any + other module that adds headers), there'll be dups in the message. + + ASTERISK-25337 #close + + Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa + (cherry picked from commit 34c64707d1aa346fb0e9c7f97e375d22dedf67d9) + +2015-05-02 02:15 +0000 [23554d74ab] Corey Farrell + + * res_pjsip_dlg_options: Fix MODULEINFO section. + + Removed the extra space before "MODULEINFO" in res_pjsip_dlg_options. + This extra space prevented any of the dependencies from being seen by + menuselect, so building with default options would fail if PJSIP was + not installed. + + This also makes the tool that extracts information for menuselect + tolerant of multiple spaces in the future. + + ASTERISK-25033 #close + Reported by: Peter Whisker + + Change-Id: Iccd54846f70c4a7a50cb5bf70b7bb5cb4bab3698 + 2016-04-27 16:18 +0000 Asterisk Development Team * asterisk certified/13.1-cert7 Released. -- 2.47.2