+2016-07-19 15:29 +0000 Asterisk Development Team <asteriskteam@digium.com>
+
+ * asterisk certified/13.1-cert8 Released.
+
+2016-07-19 10:29 +0000 [6837d582bc] Mark Michelson <mmichelson@digium.com>
+
+ * Release summaries: Remove previous versions
+
+2016-07-19 10:29 +0000 [81bda18275] Mark Michelson <mmichelson@digium.com>
+
+ * .version: Update for certified/13.1-cert8
+
+2016-07-19 10:29 +0000 [8cfe7d9aac] Mark Michelson <mmichelson@digium.com>
+
+ * .lastclean: Update for certified/13.1-cert8
+
+2016-07-19 10:29 +0000 [cca43b1a2f] Mark Michelson <mmichelson@digium.com>
+
+ * realtime: Add database scripts for certified/13.1-cert8
+
+2016-07-07 10:38 +0000 [690e23433b] Joshua Colp <jcolp@digium.com>
+
+ * 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 <rmudgett@digium.com>
+
+ * 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 <gjoseph@digium.com>
+
+ * 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 <gjoseph@digium.com>
+
+ * 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 <jcolp@digium.com>
+
+ * 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 <george.joseph@fairview5.com>
+
+ * 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 <git@cfware.com>
+
+ * 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 <asteriskteam@digium.com>
* asterisk certified/13.1-cert7 Released.
projects / thirdparty / asterisk.git / commitdiff
