asterisk

Commit Graph

Author	SHA1	Message	Date
Olle Johansson	ede3699c6e	Merged revisions 227088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r227088 \| oej \| 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) \| 7 lines Use proper response code when violating Contact ACL's. https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	66579d9d49	Add PacketCable NCS 1.0 support for Docsis/Eurodocsis networks (closes issue #12950) Reported by: alea-soluciones Patches: ncs-pktccops-12950-r206803.patch uploaded by alea-soluciones (license 514) Tested by: alea-soluciones, adomjan, urtho, nahuelgreco git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	2c4d3b3168	SIP channel name uniqueness SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	6406f39594	DAHDI ISDN channel names will not allow device state to work. (Interim solution.) Since ISDN works like SIP and not analog ports in regard to devices, the device state based on the ISDN channel number could not work. This has not been an issue until the advent of PTMP NT mode. Previously, ISDN lines were used as trunks and did not have to keep track of specific devices. As an interim solution until device states are properly implemented, the channel name is being changed to the following format to use the generic device state support: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> Dialplan hints would thus be: exten => xxx,hint,DAHDI/i2/5551212 This will work with the following restrictions: * The number of devices/phones cannot exceed the number of B channels. (i.e., BRI has 2) * Each device/phone can only have one number. No shared MSN's. * The phones/devices probably should not use subaddressing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	93e43578ec	This patch adds support for a draft proposal for adding Q.850 reason headers to sip messages. (closes issue #13385) Reported by: adomjan Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by adomjan (license 487) sip-q850-hangupcause1.diff uploaded by mnicholson (license 96) Tested by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	7fbd314a88	Cleanup some flags on DAHDI PRI channel hangup. * Cleanup some flags on DAHDI PRI channel hangup. (sig_pri split) * Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls. * Remove some unused flags since sig_pri was split. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	b9c370da86	Merged revisions 226531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226531 \| file \| 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) \| 6 lines Add an option to enabling passing music on hold start and stop requests through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	64e8fb465b	Doxygen documentation update git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	5825f68e8b	Add support for receiving unsolicited MWI NOTIFY messages. This change adds a configuration option to SIP peers, unsolicited_mailbox, which configures a virtual mailbox to use for received new/old MWI information. This virtual mailbox can then be used by any device supporting MWI. (closes issue #13028) Reported by: AsteriskRocks Patches: bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ea8b54fb9d	Fix building in REF_DEBUG mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	ec0a1882c9	ACL check not present for verifying SIP INVITEs The ACL check in check_peer_ok was missing and has now been restored. The missing check allowed for calls to be made on prohibited networks where an ACL was defined in sip.conf and the allowguest option was set to off. See the AST security advisory below for more information. Merge code associated with AST-2009-007. (closes issue #16091) Reported by: thom4fun git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	71452322a2	Make conditionals create previous code when libpri/ss7 are present. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	2736168a6b	span numbers in pri debug / error messages Prefix PRI trace messages with the span number. This makes the trace readable even when you have a multi-port device. (closes issue #15054) Reported by: tzafrir Patches: dahdi_pri_debug_spannum.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	e5a57959eb	Re-arange code a bit to build in dev-mode without ss7 No change of functionality here. Just localized a variable and indented code into blocks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	d36cecd578	Make chan_dahdi build even without PRI / SS7 (Note: still some strange build warnings without SS7 in dev-mode) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	fb0196fce6	Improve performance of pedantic mode dialog searching in chan_sip. This patch changes chan_sip to use the new astobj2 OBJ_MULTIPLE iterator support to make pedantic mode dialog searching in find_call() not require a linear search of all dialogs in the list of dialogs. This patch does not change the dialog matching logic (more on that later), just improves the searching performance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	cff6d02b53	Add to chan_dahdi ISDN HOLD, Call deflection, and keypad facility support. * Added handling of received HOLD/RETRIEVE messages and the optional ability to transfer a held call on disconnect similar to an analog phone. * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP. Will reroute/deflect an outgoing call when receive the message. Can use the DAHDISendCallreroutingFacility to send the message for the supported switches. * Added ability to send/receive keypad digits in the SETUP message. Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension]) Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)} * Added support for BRI PTMP NT mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	2208fb171b	Fixes an iterator memory leak and uninitialized memory git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225650 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	63473616da	Search for the subaddress only within the extension section of the dial string. Dial(DAHDI/(g\|G\|r\|R)<group#(0-63)>[c\|r<cadance#>\|d][/extension]) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	776a14386a	SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes. What this patch fixes 1.Moves sip TCP/TLS connection setup into the TCP helper thread: Connection setup takes awhile and before this it was being done while holding the monitor lock. 2.Moves TCP/TLS writing to the TCP helper thread: Through the use of a packet queue and an alert pipe, the TCP helper thread can now be woken up to write data as well as read data. 3.Locking error: sip_xmit returned an XMIT_ERROR without giving up the tcptls_session lock. This lock has been completely removed from sip_xmit and placed in the new sip_tcptls_write() function. 4.Memory leak: When creating a tcptls_client the tls_cfg was alloced but never freed unless the tcptls_session failed to start. Now the session_args for a sip client are an ao2 object which frees the tls_cfg on destruction. 5.Pointer to stack variable: During sip_prepare_socket the creation of a client's ast_tcptls_session_args was done on the stack and stored as a pointer in the newly created tcptls_session. Depending on the events that followed, there was a slight possibility that pointer could have been accessed after the stack returned. Given the new changes, it is always accessed after the stack returns which is why I found it. Notable code changes 1.I broke tcptls.c's ast_tcptls_client_start() function into two functions. One for creating and allocating the new tcptls_session, and a separate one for starting and handling the new connection. This allowed me to create the tcptls_session, launch the helper thread, and then establish the connection within the helper thread. 2.Writes to a tcptls_session are now done within the helper thread. This is done by using an alert pipe to wake up the thread if new data needs to be sent. The thread's sip_threadinfo object contains the alert pipe as well as the packet queue. 3.Since the threadinfo object contains the alert pipe, it must now be accessed outside of the helper thread for every write (queuing of a packet). For easy lookup, I moved the threadinfo objects from a linked list to an ao2_container. (closes issue #13136) Reported by: pabelanger Tested by: dvossel, whys (closes issue #15894) Reported by: dvossel Tested by: dvossel Review: https://reviewboard.asterisk.org/r/380/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	1174a61612	Add support for calling and called subaddress. Partial support for COLP subaddress. The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing "desk to desk" between offices each with an asterisk box over the ISDN should then be possible, without a whole load of DDI numbers required. (closes issue #15604) Reported by: alecdavis Patches: asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585) Some minor modificatons were made. Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/405/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3acfd4933c	Merged revisions 225243 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225243 \| dvossel \| 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) \| 13 lines IAX2: VNAK loop caused by signaling frames with no destination call number It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	87ff40d3f3	Add 'mohsuggest' configuration option to 'sip show peer' CLI command and SIPShowPeer AMI action. (closes issue #15990) Reported by: _brent_ Patches: sip_peer_info_mohsuggest-r3.patch uploaded by brent (license 388) Review: https://reviewboard.asterisk.org/r/381/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	01ab66275a	Add support for specifying the IP address to use for media streams in sip.conf This is the second commit for this and documents the text stream using the configured IP address and fixes a bug in the original patch where the UDPTL stream would also use the different IP address. (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	a31eb5bb35	Revert media_address commit, I'm going to roll a fix to the SDP generation in the next version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	984d6500ce	Merged revisions 225032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 \| dvossel \| 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) \| 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	28d0ec5421	Add support for specifying the IP address to use for media streams in sip.conf (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	c5b8e9af7c	Make PRI_SUBCMD_xxx handling subaddress friendly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	26638d3a55	Add dynamic range compression support for analog channels. (closes issue AST-29) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	fd9f9ab01e	Add a callback to sig_pri which is called when sig_pri is going to queue a control frame on a channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	03db5ef0e5	fix typo, sorry git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	53a95d9c84	Merged revisions 224330 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 \| jpeeler \| 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) \| 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224331 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	64a32b3ad0	Merged revisions 224260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 \| rmudgett \| 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) \| 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (issue #14292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	e3f473f4f3	Allow for adding message body to the SIP NOTIFY message Ability has been added to both manager command SIPnotify as well as console command sip notify. Message body is stored in the "Content" variable. An example is present in sip_notify.conf. (closes issue #13926) Reported by: jthurman Patches: sip-notify-svn189463.diff uploaded by gareth (license 208) Tested by: gareth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e197f85b8c	Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	9e1598b762	Check the proper page for the SENDRPID flag. If a pending reinvite were sent, we might not properly send connected party info since we were checking the wrong flag. This was a rare occurrence, but could still happen nevertheless. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	fc27da108d	Merged revisions 223205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223205 \| dvossel \| 2009-10-09 12:52:35 -0500 (Fri, 09 Oct 2009) \| 10 lines fixes sip registration using authuser in user.conf (closes issue #14954) Reported by: tornblad Tested by: mmichelson, tornblad, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	0a50c512da	'auth=' did not parse md5 secret correctly (closes issue #15949) Reported by: ebroad Patches: authparsefix.patch uploaded by ebroad (license 878) 15949_trunk.diff uploaded by dvossel (license 671) Tested by: ebroad git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	b2a7eecd6e	p->peerauth is always empty in transmit_register() When using callbackextension or specifing the peer name in a registration string, the peer's specific auth settings set by the "auth=" strings within the peer definition are not used by the registration. Thanks to ebroad for reporting the issue and providing the patch. (closes issue #15955) Reported by: ebroad Patches: regauthfix.patch uploaded by ebroad (license 878) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	799e9962b6	fixed comment line for do_magic_pickup git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	c0ee60419d	Deadlock between ast_cel_report_event and ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner channel while only the pvt lock is held. Since pbx_exec calls ast_cel_report_event which attempts to lock the channel, invalid locking order occurs. Channels should be locked before pvt's. (closes issue #15512) Reported by: lmsteffan Patches: ast_cel_deadlock_15512.diff uploaded by dvossel (license 671) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222981 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	b764544641	makes externtcpport and externtlsport static variables externtcpport and externtlsport need to be declared as static variables. Thanks to russell for finding and pointing this out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	890d500287	Merged revisions 222797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222797 \| rmudgett \| 2009-10-08 11:33:06 -0500 (Thu, 08 Oct 2009) \| 12 lines Fix memory leak if chan_misdn config parameter is repeated. Memory leak when the same config option is set more than once in an misdn.conf section. Why must this be considered? Templates! Defining a template with default port options and later adding to or overriding some of them. Patches: memleak-misdn.patch JIRA ABE-1998 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	9456ab2724	Deadlock in channel masquerade handling Channels are stored in an ao2_container. When accessing an item within an ao2_container the proper locking order is to first lock the container, and then the items within it. In ast_do_masquerade both the clone and original channel must be locked for the entire duration of the function. The problem with this is that it attemptes to unlink and link these channels back into the ao2_container when one of the channel's name changes. This is invalid locking order as the process of unlinking and linking will lock the ao2_container while the channels are locked!!! Now, both the channels in do_masquerade are unlinked from the ao2_container and then locked for the entire function. At the end of the function both channels are unlocked and linked back into the container with their new names as hash values. This new method of requiring all channels and tech pvts to be unlocked before ast_do_masquerade() or ast_change_name() required several changes throughout the code base. (closes issue #15911) Reported by: russell Patches: masq_deadlock_trunk.diff uploaded by dvossel (license 671) Tested by: dvossel, atis (closes issue #15618) Reported by: lmsteffan Patches: deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671) Tested by: lmsteffan, dvossel Review: https://reviewboard.asterisk.org/r/387/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	49b90d5e61	Merged revisions 222691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222691 \| rmudgett \| 2009-10-07 16:51:24 -0500 (Wed, 07 Oct 2009) \| 14 lines chan_misdn.c:process_ast_dsp() memory leak misdn.conf: astdtmf must be set to "yes". With "no", buffer loss does not occur. The translated frame "f2" when passing through ast_dsp_process() is not freed whenever it is not used further in process_ast_dsp(). Then in the end it is never ever freed. Patches: translate.patch JIRA ABE-1993 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	4ae6bee6da	Change ringt (ring timeout) styles to be consistent across chan_dahdi. (closes issue #15684) Reported by: alecdavis Patches: chan_dahdi.bug15684.diff2.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f819ce5b20	Merged revisions 222542 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222542 \| dvossel \| 2009-10-07 12:41:21 -0500 (Wed, 07 Oct 2009) \| 8 lines crash on transfer handle_invite_replaces() attempts to uplock a pvt's owner channel without first verifing that it exists. (issue #16027) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b5eb0449c0	Merged revisions 222462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222462 \| jpeeler \| 2009-10-06 18:51:19 -0500 (Tue, 06 Oct 2009) \| 8 lines Add missing unlock(s) in dahdi_read (two cases in trunk) (closes issue #15683) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	1d40faebac	contact header port ignored transport when using externip This patch adds support for TCP/TLS in the Contact header when using NAT, specifically externip or externhost. The original issue was that Asterisk sent 5060 as the port in the contact header whether TLS was used or not. Additionally, this patch adds 2 config options to sip.conf, specifically externtcpport and externtlsport. This allows a user to specify different external ports for TCP and TLS other than those used internally, this is especially useful in in a PAT/port redirection setup. Thanks to ebroad for reporting the issue and providing the patch! (closes issue #15880) Reported by: ebroad Patches: portmap.patch uploaded by ebroad (license 878) externtXXport_v2.patch uploaded by ebroad (license 878) Tested by: ebroad Review: https://reviewboard.asterisk.org/r/392/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222398 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	f7fa417130	Fix 222298 (crash during destruction of second channel when variable set with setvar). I mistakenly reasoned that setvar would be used on all channels. Since it can be set per channel, give each dahdi channel a copy of the variable. (related to #15899) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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6291 Commits (c900eaa7334c7d92d364badefb8d4ca569f81bb5)