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2009-09-16Merged revisions 218933 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r218933 | mmichelson | 2009-09-16 14:25:36 -0500 (Wed, 16 Sep 2009) | 12 lines Reverse order of args to fread. This way, we don't always write a null byte into byte 1 of the buffer (closes issue #15905) Reported by: ebroad Patches: freadfix.patch uploaded by ebroad (license 878) Tested by: ebroad ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@218936 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-16Merged revisions 218918 via svnmerge from file1-10/+13
https://origsvn.digium.com/svn/asterisk/trunk ........ r218918 | file | 2009-09-16 13:31:47 -0500 (Wed, 16 Sep 2009) | 5 lines On TCP and TLS connections do not attempt to stop retransmission of the packet internally. This was preventing responses from being properly processed because the packet was not being found causing handle_response to return prematurely. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@218932 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 139281,175058,175089 via svnmerge from tilghman1-5/+52
https://origsvn.digium.com/svn/asterisk/trunk (closes issue #13985) ................ r139281 | phsultan | 2008-08-21 04:55:31 -0500 (Thu, 21 Aug 2008) | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel! (closes issue #13310) Reported by: eliel Patches: chan_gtalk.c.patch uploaded by eliel (license 64) ................ r175058 | phsultan | 2009-02-12 04:31:36 -0600 (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 ........ ................ r175089 | phsultan | 2009-02-12 08:25:03 -0600 (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/1.6.1@218727 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218687 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r218687 | dvossel | 2009-09-15 14:22:37 -0500 (Tue, 15 Sep 2009) | 2 lines upward bound checking for port string to int conversion ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@218689 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218586 via svnmerge from mnicholson1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r218586 | mnicholson | 2009-09-15 11:15:02 -0500 (Tue, 15 Sep 2009) | 15 lines Merged revisions 218578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218578 | mnicholson | 2009-09-15 11:03:54 -0500 (Tue, 15 Sep 2009) | 8 lines Send request contact header field with response to registrer queries instead of the address of record. (closes issue #14438) Reported by: ravindrad Patches: regquerypatch uploaded by ravindrad (license 684) Tested by: ravindrad ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@218592 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218566 via svnmerge from mmichelson1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r218566 | mmichelson | 2009-09-15 10:40:14 -0500 (Tue, 15 Sep 2009) | 4 lines Use a better method of ensuring null-termination of the buffer while reading the SDP when using TCP. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@218574 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218430 via svnmerge from jpeeler1-24/+24
https://origsvn.digium.com/svn/asterisk/trunk ................ r218430 | jpeeler | 2009-09-14 17:38:25 -0500 (Mon, 14 Sep 2009) | 18 lines Merged revisions 218401 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor. After talking to rmudgett about some of his recent iflist locking changes, it was determined that the only place that would destroy a channel without being explicitly to do so was in handle_init_event. The loop to walk the interface list has been modified to wait to destroy the channel until the dahdi_pvt of the channel to be destroyed is no longer needed. (closes issue #15378) Reported by: samy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@218569 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-15Merged revisions 218499,218504 via svnmerge from mmichelson1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r218499 | mmichelson | 2009-09-15 09:59:50 -0500 (Tue, 15 Sep 2009) | 3 lines Fix off-by-one error when reading SDP sent over TCP. ........ r218504 | mmichelson | 2009-09-15 10:05:53 -0500 (Tue, 15 Sep 2009) | 3 lines Ensure that SDP read from TCP socket is null-terminated. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@218506 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-13gcc 4.4: Remove a nop memset size 0 that annoys gcctzafrir1-1/+0
This memset doesn't write beyond the end of the buffer. (tmpbuf has size of 4). Merged revisions 218184 via svnmerge from http://svn.digium.com/svn/asterisk/trunk git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@218218 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-10Merged revisions 217916 via svnmerge from tilghman2-4/+9
https://origsvn.digium.com/svn/asterisk/trunk ........ r217916 | tilghman | 2009-09-10 18:12:16 -0500 (Thu, 10 Sep 2009) | 2 lines Make calltoken support work with realtime users and peers. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@217924 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-10Merged revisions 217807 via svnmerge from dvossel1-4/+26
https://origsvn.digium.com/svn/asterisk/trunk ................ r217807 | dvossel | 2009-09-10 16:07:47 -0500 (Thu, 10 Sep 2009) | 28 lines Merged revisions 217806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call Token security patch inadvertently broke the use of encryption due to the reorganization of code in the socket_process() function. When encryption is used, an incoming full frame must first be decrypted before the information elements can be parsed. The security release mistakenly moved IE parsing before decryption in order to process the new Call Token IE. To resolve this, decryption of full frames is once again done before looking into the frame. This involves searching for an existing callno, checking the pvt to see if encryption is turned on, and decrypting the packet before the internal fields of the full frame are accessed. (closes issue #15834) Reported by: karesmakro Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, karesmakro Review: https://reviewboard.asterisk.org/r/355/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@217826 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-10Merged revisions 217593 via svnmerge from oej1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r217593 | oej | 2009-09-10 14:06:55 +0200 (Tor, 10 Sep 2009) | 8 lines Include ActionID in all events that are responsed to AMI Action SIPShowRegistry (closes issue #15868) Reported by: nic_bellamy Patches: manager_SIPshowregistry_actionid.patch uploaded by nic bellamy (license 299) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@217595 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-09Merged revisions 217368 via svnmerge from oej1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r217368 | oej | 2009-09-09 12:39:43 +0200 (Ons, 09 Sep 2009) | 2 lines Not having any TLS session to write to is a serious XMIT_ERROR. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@217370 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-08Merged revisions 216993 via svnmerge from dvossel1-4/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r216993 | dvossel | 2009-09-08 09:26:30 -0500 (Tue, 08 Sep 2009) | 14 lines caller id number empty parse_uri was not being given the correct scheme's, as a result, uri parsing did not parse the username correctly. One of the side effects of this is an empty caller id. (closes issue #15839) Reported by: ebroad Patches: blank_cidv2.patch uploaded by ebroad (license 878) parse_uri_fix.diff uploaded by dvossel (license 671) Tested by: ebroad, dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216995 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Merged revisions 216842 via svnmerge from oej1-2/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r216842 | oej | 2009-09-07 18:35:12 +0200 (MÃ¥n, 07 Sep 2009) | 2 lines Make sure we reset global_exclude_static at channel reload ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216844 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Merged revisions 216695 via svnmerge from oej1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r216695 | oej | 2009-09-07 15:06:19 +0200 (MÃ¥n, 07 Sep 2009) | 8 lines If there is no session timer setting in the INVITE, set it to default value (not unset minimum = -1) Patch by oej closes issue #15621 Reported by: fnordian Tested by: atis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216697 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Turning off premature media by defaultoej1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216653 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Merged revisions 216438 via svnmerge from oej1-3/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines Merged revisions 216430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216646 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-04Merged revisions 216594 via svnmerge from dvossel1-50/+34
https://origsvn.digium.com/svn/asterisk/trunk ........ r216594 | dvossel | 2009-09-04 14:32:07 -0500 (Fri, 04 Sep 2009) | 7 lines sip peer matching by address only with TCP/TLS This patch removes the contact header matching logic and adds logic to match all tcp/tls connections by ip only Review: https://reviewboard.asterisk.org/r/354/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216599 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-04Merged revisions 216368 via svnmerge from russell1-8/+25
https://origsvn.digium.com/svn/asterisk/trunk ........ r216368 | russell | 2009-09-04 08:14:25 -0500 (Fri, 04 Sep 2009) | 12 lines Do not treat every SIP peer as if they were configured with insecure=port. There was a problem in the function responsible for doing peer matching by IP address and port number such that during the second pass for checking for a peer configured with insecure=port, it would end up treating every peer as if it had been configured that way. These changes fix the logic in the peer IP and port comparison callback to handle insecure=port checking properly. This problem was introduced when SIP peers were converted to astobj2. Many thanks to dvossel for noticing this while working on another peer matching issue. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216434 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03Merged revisions 215955 via svnmerge from dvossel4-145/+1227
https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216004 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03Merged revisions 215891 via svnmerge from oej1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r215891 | oej | 2009-09-03 15:02:41 +0200 (Tor, 03 Sep 2009) | 10 lines Add known internal IP address when autodomain=yes (closes issue #14573) Reported by: pj Patches: sip-internip-autodomain1.diff uploaded by mnicholson (license 96) modified by oej Tested by: pj ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@215932 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03Merged revisions 215758 via svnmerge from twilson1-12/+78
https://origsvn.digium.com/svn/asterisk/trunk ................ r215758 | twilson | 2009-09-02 18:31:04 -0500 (Wed, 02 Sep 2009) | 25 lines Merged revisions 215682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines Re-send non-100 provisional responses to prevent cancellation From section 13.3.1.1 of RFC 3261: If the UAS desires an extended period of time to answer the INVITE, it will need to ask for an "extension" in order to prevent proxies from canceling the transaction. A proxy has the option of canceling a transaction when there is a gap of 3 minutes between responses in a transaction. To prevent cancellation, the UAS MUST send a non-100 provisional response at every minute, to handle the possibility of lost provisional responses. (closes issue #11157) Reported by: rjain Tested by: twilson Review: https://reviewboard.asterisk.org/r/315/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@215774 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-02Merged revisions 215681 via svnmerge from dvossel1-21/+29
https://origsvn.digium.com/svn/asterisk/trunk ........ r215681 | dvossel | 2009-09-02 16:39:31 -0500 (Wed, 02 Sep 2009) | 10 lines port string to int conversion using sscanf There are several instances where a port is parsed from a uri or some other source and converted to an int value using atoi(), if for some reason the port string is empty, then a standard port is used. This logic is used over and over, so I created a function to handle it in a safer way using sscanf(). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@215684 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-02Merged revisions 215665 via svnmerge from mvanbaak2-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r215665 | mvanbaak | 2009-09-02 23:23:17 +0200 (Wed, 02 Sep 2009) | 5 lines add Parkinglot info to sip show peer <foo> and skinny show line <foo> If we had this from the start, debugging the 'parking not using configured parkinglot' bug would have been easier. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@215679 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-02Merged revisions 215522 via svnmerge from dvossel1-43/+40
https://origsvn.digium.com/svn/asterisk/trunk ........ r215522 | dvossel | 2009-09-02 12:26:40 -0500 (Wed, 02 Sep 2009) | 11 lines SIP uri parsing cleanup Now, the scheme passed to parse_uri can either be a single scheme, or a list of schemes ',' delimited. This gets rid of the whole problem of having to create two buffers and calling parse_uri twice to check for separate schemes. Review: https://reviewboard.asterisk.org/r/343/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@215524 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-02Merged revisions 215479 via svnmerge from mvanbaak1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r215479 | mvanbaak | 2009-09-02 18:20:23 +0200 (Wed, 02 Sep 2009) | 3 lines like in chan_sip's sip_new skinny should copy the configured parkinglot from a line to the newly created channel. This makes callparking honor the configured parkinglot for skinny lines as well. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@215511 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-02Merged revisions 215462 via svnmerge from mvanbaak1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r215462 | mvanbaak | 2009-09-02 17:56:46 +0200 (Wed, 02 Sep 2009) | 12 lines Honor configured parkinglot when parking and retrieving parked calls Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer into the newly created channel. (closes issue #15538) Reported by: gracedman Patches: 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7) With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call) Tested by: gracedman, mvanbaak ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@215464 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-31Merged revisions 214945 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r214945 | tilghman | 2009-08-31 11:18:33 -0500 (Mon, 31 Aug 2009) | 14 lines Merged revisions 214940 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009) | 7 lines Also unlock the "other" channel, when returning, due to glare. (closes issue #15787) Reported by: tim_ringenbach Patches: chan_local.diff uploaded by tim ringenbach (license 540) Tested by: tim_ringenbach ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@214958 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-26Merged revisions 214199 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r214199 | tilghman | 2009-08-26 11:53:03 -0500 (Wed, 26 Aug 2009) | 6 lines Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) (closes issue #15362) Reported by: klaus3000 Patches: chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license 65) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@214201 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-21Merged revisions 213716 via svnmerge from dvossel1-14/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r213716 | dvossel | 2009-08-21 17:22:11 -0500 (Fri, 21 Aug 2009) | 10 lines Register request line contains wrong address when user domain and register host differ (closes issue #15539) Reported by: Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@213724 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-19Merged revisions 213093 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r213093 | tilghman | 2009-08-19 15:29:41 -0500 (Wed, 19 Aug 2009) | 7 lines If we have realtime caching enabled, 'sip reload' must purge users/peers, even if the config files haven't changed. (closes issue #12869) Reported by: bcnit Patches: 20090819__issue12869__2.diff.txt uploaded by tilghman (license 14) Tested by: lasko ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@213096 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-18Merged revisions 212758 via svnmerge from rmudgett1-29/+30
https://origsvn.digium.com/svn/asterisk/trunk ................ r212758 | rmudgett | 2009-08-18 11:29:47 -0500 (Tue, 18 Aug 2009) | 9 lines Merged revisions 212727 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 Aug 2009) | 1 line Removed some deadwood and added some doxygen comments. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@212768 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-17Merged revisions 212581 via svnmerge from seanbright1-1/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r212581 | seanbright | 2009-08-17 14:50:24 -0400 (Mon, 17 Aug 2009) | 5 lines Correct spelling of AGENTACCEPTDTMF in chan_agent. (closes issue #15668) Reported by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@212583 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-17Merged revisions 212506 via svnmerge from jpeeler1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r212506 | jpeeler | 2009-08-17 11:50:45 -0500 (Mon, 17 Aug 2009) | 19 lines Merged revisions 212498 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009) | 12 lines Fix segfault when reloading chan_misdn. If more ports were specified than configured in misdn.conf a reload would crash asterisk. The problem was the unconfigured port was using data from the previously configured port. When the data for an unconfigured port was freed a crash would result from the double free. (closes issue #12113) Reported by: agupta Patches: bug12113.patch uploaded by jpeeler (license 325) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@212508 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-17Merged revisions 212431 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r212431 | rmudgett | 2009-08-17 10:42:51 -0500 (Mon, 17 Aug 2009) | 16 lines Merged revisions 212430 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix uninitialized variable causing random MWI indications. (closes issue #15727) Reported by: doda Patches: dahdi_changes.patch uploaded by doda (license 853) ........ r212430 | rmudgett | 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix uninitialized variable. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@212433 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-13Merged revisions 212113 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r212113 | kpfleming | 2009-08-13 10:46:25 -0500 (Thu, 13 Aug 2009) | 3 lines Ensure that T38FaxVersion is put into outgoing SDP in the proper case. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@212115 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-13Merged revisions 212067 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r212067 | file | 2009-08-13 10:51:04 -0300 (Thu, 13 Aug 2009) | 2 lines Check an actual populated variable when seeing if we need to do video or not. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@212069 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-12Merged revisions 211876 via svnmerge from mnicholson1-3/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r211876 | mnicholson | 2009-08-12 14:53:14 -0500 (Wed, 12 Aug 2009) | 11 lines Make asterisk handle 423 Interval Too Short messages better. This change uses separate values for the acceptable minimum expiry provided by the 423 error and the expiry value stored in the configuration file. Previously, the value pulled from the configuration file would be overwritten. (closes issue #14366) Reported by: Nick_Lewis Patches: sip-expiry-fix1.diff uploaded by mnicholson (license 96) chan_sip.c-reqexpiry.patch uploaded by Nick (license 657) Tested by: mnicholson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@211950 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman12-126/+126
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@211569 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10Merged revisions 211347 via svnmerge from file1-10/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 | file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix retrieval of the port used for the video stream when adding SDP to a SIP message. (closes issue #15121) Reported by: jsmith ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@211349 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-06Merged revisions 210817 via svnmerge from file1-91/+93
https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 | file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines Accept additional T.38 reinvites after an initial one has been handled. Discussion of this subject has yielded that it is not actually acceptable to change T.38 parameters after the initial reinvite but declining is harsh and can cause the fax to fail when it may be possible to allow it to continue. This patch changes things so that additional T.38 reinvites are accepted but parameter changes ignored. This gives the fax a fighting chance. (closes issue #15610) Reported by: huangtx2009 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@210819 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-05Merged revisions 210640 via svnmerge from rmudgett1-49/+62
https://origsvn.digium.com/svn/asterisk/trunk ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500 (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009) | 14 lines Dialplan starts execution before the channel setup is complete. * Issue 15655: For the case where dialing is complete for an incoming call, dahdi_new() was asked to start the PBX and then the code set more channel variables. If the dialplan hungup before these channel variables got set, asterisk would likely crash. * Fixed potential for overlap incoming call to erroneously set channel variables as global dialplan variables if the ast_channel structure failed to get allocated. * Added missing set of CALLINGSUBADDR in the dialing is complete case. (closes issue #15655) Reported by: alecdavis ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@210681 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-01Merged revisions 209760-209761 via svnmerge from kpfleming2-6/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500 (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul 2009) | 7 lines Minor changes inspired by testing with latest GCC. The latest GCC (what will become 4.5.x) has a few new warnings, that in these cases found some either downright buggy code, or at least seriously poorly designed code that could be improved. ........ ................ r209761 | kpfleming | 2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert accidental Makefile change. ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@209781 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-30Merged revisions 209554 via svnmerge from dbrooks2-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@209593 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-30Merged revisions 209516 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r209516 | mmichelson | 2009-07-30 09:38:21 -0500 (Thu, 30 Jul 2009) | 8 lines Fix a crash that can result if text codecs are allowed but textsupport is disabled. (closes issue #15596) Reported by: fabled Patches: sip-red.patch uploaded by fabled (license 448) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@209517 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-27Merged revisions 209098 via svnmerge from dbrooks2-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@209233 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-27Merged revisions 208924 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009) | 9 lines Merged revisions 208923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009) | 2 lines Fix logic errors from 208746 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@208926 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-25Merged revisions 208749 via svnmerge from jpeeler2-4/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009) | 13 lines Merged revisions 208746 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009) | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly trivial changes, but I did not know of any other way to fix the "dereferencing type-punned pointer will break strict-aliasing rules" error without creating a tmp variable in chan_skinny. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@208754 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-24Merged revisions 208588 via svnmerge from mmichelson1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul 2009) | 16 lines Merged revisions 208587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul 2009) | 10 lines Only send a BYE when hanging up a channel that is up. For cases where Asterisk sends an INVITE and receives a non 2XX final response, Asterisk would follow the INVITE transaction by immediately sending a BYE, which was unnecessary. (closes issue #14575) Reported by: chris-mac ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@208590 f38db490-d61c-443f-a65b-d21fe96a405b
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