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2009-03-19Merged revisions 183108 via svnmerge from file1-181/+34
https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines Improve our triggering of a T38 switchover internally when triggered by a received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@183109 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Merged revisions 183028 via svnmerge from jpeeler1-1/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@183029 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Merged revisions 182847 via svnmerge from russell2-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182945 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Merged revisions 182722 via svnmerge from jpeeler6-75/+126
https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182723 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-16Merged revisions 182282 via svnmerge from dvossel1-1/+18
https://origsvn.digium.com/svn/asterisk/trunk ................ r182282 | dvossel | 2009-03-16 12:49:58 -0500 (Mon, 16 Mar 2009) | 13 lines Merged revisions 182281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182283 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-16Merged revisions 182211 via svnmerge from tilghman1-11/+16
https://origsvn.digium.com/svn/asterisk/trunk ................ r182211 | tilghman | 2009-03-16 10:50:55 -0500 (Mon, 16 Mar 2009) | 14 lines Merged revisions 182208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182212 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-13Merged revisions 182022 via svnmerge from file1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r182022 | file | 2009-03-13 14:25:09 -0300 (Fri, 13 Mar 2009) | 7 lines Fix an issue with requesting a T38 reinvite before the call is answered. The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182036 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181769 via svnmerge from mmichelson1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r181769 | mmichelson | 2009-03-12 13:30:58 -0500 (Thu, 12 Mar 2009) | 28 lines Merged revisions 181768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181770 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181371 via svnmerge from dvossel2-31/+71
https://origsvn.digium.com/svn/asterisk/trunk ................ r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181372 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181345 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines Merged revisions 181328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181352 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181296 via svnmerge from file1-3/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines Merged revisions 181295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181297 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11add missing header filejpeeler1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181284 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181135 via svnmerge from jpeeler2-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181137 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181032-181033 via svnmerge from mmichelson1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500 (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines Remove unused variables. ........ ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500 (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC 3891 ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181034 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10If a port is specified when dialing a peer then use it.file1-0/+5
(closes issue #14626) Reported by: acunningham git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180799 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Ensure that the new outgoing dialog to a peer is able to set the socket ↵file1-7/+11
details, even if the default is present. (closes issue #14480) Reported by: jon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180718 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 151464 via svnmerge from mmichelson1-12/+21
https://origsvn.digium.com/svn/asterisk/trunk ........ r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines Make the sip_standard_port function more granular by allowing separate type and port arguments. This is necessary because when building our From and Contact headers, we need to be absolutely sure that we are placing our source port there and not the peer's source port. (closes issue #12761) Reported by: asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179473 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-01Merged revisions 179219 via svnmerge from mmichelson1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines Properly free memory and remove scheduler entries when a transmission failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_FAILURE is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179220 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Merged revisions 178871 via svnmerge from dvossel1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) | 6 lines IAX2 prune realtime, minor tweak to last fix A return statement was missing which caused unexpected cli output. issue #14479 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178874 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Merged revisions 178767 via svnmerge from dvossel1-16/+43
https://origsvn.digium.com/svn/asterisk/trunk ........ r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178768 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-24Merged revisions 178213 via svnmerge from file1-3/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) | 16 lines Merged revisions 178205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178224 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-20Fixes issue with undefined audio codecs in chan_iax2dvossel1-1/+1
During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.6 only audio codec bits 0-12 and 15 are defined, leaving bits 13-14 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-14 are not defined, these bits are never turned off. In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue #14283) Reported by: jcovert git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177698 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-19Remove invalid ast_free calls for static character arraysjpeeler1-6/+0
(issue #14478) Reported by: erik_dedecker git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177445 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 177162 via svnmerge from jpeeler10-110/+146
https://origsvn.digium.com/svn/asterisk/trunk ........ r177162 | jpeeler | 2009-02-18 14:11:57 -0600 (Wed, 18 Feb 2009) | 14 lines Modify h323 to build against PTLib as well as the older PWLib Several changes in PTLib have occurred requiring build time detection. Changes accounted for include the library name change, config option change, install location change, and a boolean type change which is handled by ast_ptlib.h. Also, the sed check has been modified to properly work with autoconf >= 2.62. (closes issue #14224) Reported by: bergolth Patches: asterisk-autoconf-sed.patch uploaded by bergolth (license 661) asterisk-pwlib-v3.patch uploaded by bergolth (license 661) Tested by: jpeeler ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177163 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 177005 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r177005 | file | 2009-02-18 13:11:52 -0400 (Wed, 18 Feb 2009) | 6 lines Fix ordering of output for a ChannelUpdate manager event. (closes issue #14497) Reported by: vinsik Patches: chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177006 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176705 via svnmerge from dhubbard1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r176705 | dhubbard | 2009-02-17 15:59:38 -0600 (Tue, 17 Feb 2009) | 11 lines create a UDPTL structure in create_addr_from_peer() if it does not already exist for T38 This is required to create a UDPTL structure in create_addr_from_peer() to handle the scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but is defined the peer's context. I tested this patch by enabling t38pt_udptl in the [general] section on one system and only enabling t38pt_udptl in a peer's context on the system sending a fax. Without the patch, the sending system will fail to initiate T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure". When this patch is applied the sending side will successfully initiate T38 negotiation. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176709 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176642 via svnmerge from tilghman1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r176642 | tilghman | 2009-02-17 15:14:18 -0600 (Tue, 17 Feb 2009) | 8 lines Prior to masquerade, move the group definitions to the channel performing the masq, so that the group count lingers past the bridge. (closes issue #14275) Reported by: kowalma Patches: 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176643 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Oops, merge broke 1.6.0.tilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176548 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17In 1.6.0, the tablename is stored in a variable.tilghman1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176502 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176459 via svnmerge from tilghman1-6/+29
https://origsvn.digium.com/svn/asterisk/trunk ................ r176459 | tilghman | 2009-02-16 19:58:39 -0600 (Mon, 16 Feb 2009) | 17 lines Merged revisions 176426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines After a 'sip reload', qualifies for realtime peers weren't immediately restarted, instead waiting until the next registration. We're now caching the qualify across a reload/restart and starting the qualify immediately upon loading the peer. (closes issue #14196) Reported by: pdf Patches: 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) Tested by: pdf ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176460 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176355 via svnmerge from dvossel1-4/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r176355 | dvossel | 2009-02-16 17:33:55 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that. issue #13749 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176359 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176248 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r176248 | dvossel | 2009-02-16 15:30:17 -0600 (Mon, 16 Feb 2009) | 11 lines Merged revisions 175597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176250 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176030 via svnmerge from file1-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r176030 | file | 2009-02-16 11:36:19 -0400 (Mon, 16 Feb 2009) | 16 lines Merged revisions 176029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176031 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 175952 via svnmerge from mvanbaak2-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r175952 | mvanbaak | 2009-02-16 01:26:59 +0100 (Mon, 16 Feb 2009) | 10 lines Merged revisions 175921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@176022 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175368 via svnmerge from russell1-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 | russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines Remove useless string copy, and make sscanf safe again ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175369 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175295 via svnmerge from russell1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r175295 | russell | 2009-02-12 14:45:47 -0600 (Thu, 12 Feb 2009) | 2 lines Avoid using ast_strdupa() in a loop. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175296 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175089 via svnmerge from phsultan1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r175089 | phsultan | 2009-02-12 15:25:03 +0100 (Thu, 12 Feb 2009) | 6 lines Issue a warning message if our candidate's IP is the loopback address. (closes issue #13985) Reported by: jcovert Tested by: phsultan ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175090 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175058 via svnmerge from phsultan1-5/+47
https://origsvn.digium.com/svn/asterisk/trunk ................ r175058 | phsultan | 2009-02-12 11:31:36 +0100 (Thu, 12 Feb 2009) | 20 lines Merged revisions 175029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@175059 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Merged revisions 174710 via svnmerge from file1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r174710 | file | 2009-02-10 16:15:43 -0400 (Tue, 10 Feb 2009) | 4 lines Only decrease inringing count if above zero. (issue #13238) Reported by: kowalma ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174711 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-10Merged revisions 174543 via svnmerge from file1-6/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r174543 | file | 2009-02-10 11:37:07 -0400 (Tue, 10 Feb 2009) | 6 lines Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported. (closes issue #14399) Reported by: caspy (issue #13238) Reported by: kowalma ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174544 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-09Merged revisions 174327 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r174327 | mmichelson | 2009-02-09 11:27:32 -0600 (Mon, 09 Feb 2009) | 3 lines Fix something I messed up in the merge I just did ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174328 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-09Merged revisions 174301 via svnmerge from mmichelson1-2/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r174301 | mmichelson | 2009-02-09 11:20:55 -0600 (Mon, 09 Feb 2009) | 20 lines Merged revisions 174282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174322 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Merged revisions 174084 via svnmerge from dhubbard1-7/+32
https://origsvn.digium.com/svn/asterisk/trunk ................ r174084 | dhubbard | 2009-02-06 17:51:56 -0600 (Fri, 06 Feb 2009) | 13 lines Merged revisions 174082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa ........ ................ ------------------------------------------------------------------------ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174085 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Merged revisions 174041 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r174041 | file | 2009-02-06 15:28:53 -0400 (Fri, 06 Feb 2009) | 4 lines Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription. (closes issue #14322) Reported by: amessina ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@174042 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Merged revisions 173974 via svnmerge from file1-42/+39
https://origsvn.digium.com/svn/asterisk/trunk ................ r173974 | file | 2009-02-06 13:18:35 -0400 (Fri, 06 Feb 2009) | 15 lines Merged revisions 173967-173968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger ........ r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a debug message I put in by accident. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173986 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-06Merged revisions 173952 via svnmerge from mnicholson1-1/+62
https://origsvn.digium.com/svn/asterisk/trunk ................ r173952 | mnicholson | 2009-02-06 10:28:19 -0600 (Fri, 06 Feb 2009) | 14 lines Merged revisions 173917 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173963 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-03Fixes issue with IAX2 transfer not handing of calls. dvossel1-8/+74
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. (issue #13468) Review: http://reviewboard.digium.com/r/140/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@173250 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 172400 via svnmerge from rmudgett1-47/+418
https://origsvn.digium.com/svn/asterisk/trunk ........ r172400 | rmudgett | 2009-01-29 14:38:34 -0600 (Thu, 29 Jan 2009) | 12 lines channels/chan_dahdi.c * Added doxygen comments to the major dahdi structures. * Fixed PRI and SS7 using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed SS7 not checking if the dialed extension is at least as long as the stripmsd option. * Fixed PRI not handling unknown TON/NPI prefix letters correctly. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172434 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-29Merged revisions 172173 via svnmerge from oej1-6/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r172173 | oej | 2009-01-29 10:18:01 +0100 (Tor, 29 Jan 2009) | 24 lines Merged revisions 172169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause. This patch implements a temporary storage in the pvt and use that instead. The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header) Thanks to Klaus Darillion for testing! (closes issue #14294) related to issue #13385 Reported by: klaus3000 and adomjan Patches: bug14294b.diff uploaded by oej (license 306) Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487) Tested by: oej, klaus3000 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@172217 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-28Merged revisions 171964 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r171964 | tilghman | 2009-01-28 11:27:40 -0600 (Wed, 28 Jan 2009) | 9 lines Merged revisions 171963 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines Clarify log message (suggested by manxpower on #asterisk-dev) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@171965 f38db490-d61c-443f-a65b-d21fe96a405b
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