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${ noResults }
2405 Commits (0101f564a4d1bcc8923e52dfbc0fc08c5f6397d5)
Author | SHA1 | Message | Date |
---|---|---|---|
|
1f96800e79 |
Merge in changes to return chan_sip to matching based on how it was previously done and how it is done in trunk. It will do name based for users and friends and IP based for peers.
(closes issue #14107) Reported by: jsmith git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@165606 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
f9973f81a3 |
Merged revisions 165216 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r165216 | file | 2008-12-17 15:52:40 -0400 (Wed, 17 Dec 2008) | 4 lines Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly! (closes issue #14055) Reported by: chris-mac ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@165218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
cbad8e6dca |
Merged revisions 164978 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r164978 | mmichelson | 2008-12-16 17:06:04 -0600 (Tue, 16 Dec 2008) | 15 lines Merged revisions 164977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec 2008) | 7 lines After looking through SIP registration code most of the day, this is one of the few things I could find that was just plain wrong. Even though it probably isn't possible for it to happen, it seems weird to have code that checks if a pointer is NULL and then immediately dereferences that pointer if it was NULL. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
072033cdb0 |
Merged revisions 164675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r164675 | russell | 2008-12-16 10:00:29 -0600 (Tue, 16 Dec 2008) | 19 lines Merged revisions 164672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines Fix a memory leak related to the use of the "setvar" configuration option. The problem was that these variables were being appended to the list of vars on the sip_pvt every time a re-registration or re-subscription came in. Since it's just a waste of memory to put them there unless the request was an INVITE, then the fix is to check the request type before copying the vars. (closes issue #14037) Reported by: marvinek Tested by: russell ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r164659 | file | 2008-12-16 11:44:28 -0400 (Tue, 16 Dec 2008) | 4 lines When using externhost make sure the port gets set to the bindaddr port if one was not specified in the externhost value itself. (closes issue #13634) Reported by: performer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164662 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 164351 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r164351 | file | 2008-12-15 14:12:24 -0400 (Mon, 15 Dec 2008) | 13 lines Merged revisions 164350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6 lines Do not try to unlock a non-existant channel if the transfer fails. (closes issue #13800) Reported by: dwagner Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164353 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 163670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r163670 | russell | 2008-12-12 12:45:03 -0600 (Fri, 12 Dec 2008) | 6 lines Rename a number of tcptls_session variables. There are no functional changes here. The name "ser" was used in a lot of places. However, it is a relic from when the struct was a server_instance, not a session_instance. It was renamed since it represents both a server or client connection. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r163667 | russell | 2008-12-12 12:33:27 -0600 (Fri, 12 Dec 2008) | 5 lines Fix a small race condition in sip_tcp_locate(). We must increase the reference count on the tcptls_session _before_ unlocking the thread list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163669 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r163642 | russell | 2008-12-12 12:19:47 -0600 (Fri, 12 Dec 2008) | 7 lines Resolve crashes when using SIP TCP/TLS with qualify. The problem was a reference count error on the tcptls_session structure. (closes issue #13989) Reported by: Nugget ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163656 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r163629 | file | 2008-12-12 14:17:12 -0400 (Fri, 12 Dec 2008) | 4 lines When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found. (closes issue #13811) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r163579 | file | 2008-12-12 12:55:15 -0400 (Fri, 12 Dec 2008) | 4 lines Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven. (closes issue #13525) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163581 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r162997 | file | 2008-12-11 11:05:49 -0400 (Thu, 11 Dec 2008) | 4 lines When a device registers to use it is entirely possible that they may be in use, so tell the core that we don't know the devstate and have it ask us for it. (closes issue #13525) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@163006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 162805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r162805 | file | 2008-12-10 15:02:57 -0400 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI. (closes issue #12560) Reported by: vsauer Patches: patch001.diff uploaded by ramonpeek (license 266) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162807 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 162739 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r162739 | file | 2008-12-10 13:53:09 -0400 (Wed, 10 Dec 2008) | 13 lines Merged revisions 162738 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0. (closes issue #13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded by hjourdain (license 583) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162741 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 162664 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r162664 | mmichelson | 2008-12-10 10:34:35 -0600 (Wed, 10 Dec 2008) | 19 lines Merged revisions 162663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec 2008) | 11 lines Revert fix for issue 13570. It has caused more problems than it helped to fix. (closes issue #13783) Reported by: navkumar (closes issue #14025) Reported by: ffs ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162666 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 162619 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r162619 | file | 2008-12-10 11:22:26 -0400 (Wed, 10 Dec 2008) | 4 lines When transmitting a register set the socket port to the local one for the transport being used, not the port for the remote server. (closes issue #13633) Reported by: performer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 161726 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r161726 | file | 2008-12-08 13:53:32 -0400 (Mon, 08 Dec 2008) | 13 lines Merged revisions 161725 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 lines Make the usereqphone option work again. (closes issue #13474) Reported by: mmaguire Patches: 20080912_bug13474.diff uploaded by mmaguire (license 571) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@161729 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r161721 | mnicholson | 2008-12-08 11:23:41 -0600 (Mon, 08 Dec 2008) | 7 lines Fix a crash that can occur on a transfer in chan_sip when attempting to collect rtp stats. (closes issue #13956) Reported by: chris-mac Tested by: chris-mac ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@161722 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 160481 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r160481 | tilghman | 2008-12-03 08:11:53 -0600 (Wed, 03 Dec 2008) | 14 lines Merged revisions 160480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I guess that having only ip-phones in mind is not a good approach. Since it is possible to have a sip proxy connected to asterisk we could receive a 407 (unauthorized) or 483 (too many hops) as response and dialog ending would not be a good behavior." So modified. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@160483 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 160308 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r160308 | tilghman | 2008-12-02 11:56:24 -0600 (Tue, 02 Dec 2008) | 17 lines Merged revisions 160297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion fails, and the resulting integer is garbage. Thus, we must initialize the integer and check it afterwards for success. (closes issue #14000) Reported by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@160325 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 159818 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r159818 | kpfleming | 2008-11-29 11:57:39 -0600 (Sat, 29 Nov 2008) | 18 lines incorporates r159808 from branches/1.4: ------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@159851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 158265-158266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r158265 | mmichelson | 2008-11-20 19:14:20 -0600 (Thu, 20 Nov 2008) | 4 lines Use some magic constants to get the right size for this sscanf statement. Thanks Richard! ........ r158266 | mmichelson | 2008-11-20 19:22:18 -0600 (Thu, 20 Nov 2008) | 3 lines Use a more expressive constant for a 64-bit scanned int ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158268 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 158262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r158262 | mmichelson | 2008-11-20 18:59:23 -0600 (Thu, 20 Nov 2008) | 6 lines Fix the build for 32-bit systems. %lu is only 32-bits on 32-bit systems, so we need to use %llu instead. Of course %llu is 128-bits on 64-bit systems, so we have to cast to unsigned long long. No harm, but it's sure annoying. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158264 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 158230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r158230 | mmichelson | 2008-11-20 17:12:50 -0600 (Thu, 20 Nov 2008) | 20 lines Change the remote user agent session version variable from an int to a uint64_t. This prevents potential comparison problems from happening if the version string exceeds INT_MAX. This was an apparent problem for one user who could not properly place a call on hold since the version in the SDP of the re-INVITE to place the call on hold greatly exceeded INT_MAX. This also aligns with RFC 2327 better since it recommends using an NTP timestamp for the version (which is a 64-bit number). (closes issue #13531) Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut (license 60) Tested by: sgofferj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158232 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Change this so it actually compiles. Thanks, Terry!
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Merged revisions 158133 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r158133 | mmichelson | 2008-11-20 12:20:00 -0600 (Thu, 20 Nov 2008) | 10 lines Merged revisions 158072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines Begin on a crusade to end trailing whitespace! ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158134 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 158082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r158082 | mmichelson | 2008-11-20 11:54:31 -0600 (Thu, 20 Nov 2008) | 24 lines Merged revisions 158071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines We don't handle 4XX responses to BYE well. According to section 15 of RFC 3261, we should terminate a dialog if we receive a 481 or 408 in response to our BYE. Since I am aware of at least one phone manufacturer who may sometimes send a 404 as well, I am being liberal and saying that any 4XX response to a BYE should result in a terminated dialog. (closes issue #12994) Reported by: pabelanger Patches: 12994.patch uploaded by putnopvut (license 60) Closes AST-129 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158084 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 158066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r158066 | mmichelson | 2008-11-20 11:39:06 -0600 (Thu, 20 Nov 2008) | 20 lines Merged revisions 158053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158067 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 157512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r157512 | mmichelson | 2008-11-18 16:54:08 -0600 (Tue, 18 Nov 2008) | 21 lines Merged revisions 157503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines Add some missing invite state changes necessary in the sip_write function. Not setting the invite state correctly on the call was resulting in the Record application leaving empty files. I also have updated the doxygen comment next to the declaration of the INV_EARLY_MEDIA constant to reflect that we also use this state when we *send* a 18X response to an INVITE. (closes issue #13878) Reported by: nahuelgreco Patches: sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162) Tested by: putnopvut ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@157531 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r157496 | mmichelson | 2008-11-18 15:59:24 -0600 (Tue, 18 Nov 2008) | 6 lines Based on Russell's advice on the asterisk-dev list, I have changed from using a global lock in update_call_counter to using the locks within the sip_pvt and sip_peer structures instead. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@157498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 157427 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r157427 | mmichelson | 2008-11-18 14:23:58 -0600 (Tue, 18 Nov 2008) | 13 lines * Add a lock to be used in the update_call_counter function. * Revert logic to mirror 1.4's in the sense that it will not allow the call counter to dip below 0. These two measures prevent potential races that could cause a SIP peer to appear to be busy forever. (closes issue #13668) Reported by: mjc Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@157429 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r157041 | russell | 2008-11-14 22:25:57 -0600 (Fri, 14 Nov 2008) | 3 lines Fix a few more places where the case insensitive hash should be used since the comparison is case insensitive. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@157042 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r156962 | mmichelson | 2008-11-14 15:19:58 -0600 (Fri, 14 Nov 2008) | 7 lines Revision 155513 of chan_sip.c in trunk inadvertently removed a very important line to set the "len" field for incoming SIP requests. The result was that all incoming SIP messages appeared to be 0-length, meaning Asterisk could do no meaningful processing of anything SIP-related ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@156963 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r155516 | russell | 2008-11-08 15:46:43 -0600 (Sat, 08 Nov 2008) | 3 lines - Check for failure when putting the packet in the ast_str - fix a spelling error in a header file ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@155517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r155513 | russell | 2008-11-08 15:34:36 -0600 (Sat, 08 Nov 2008) | 3 lines Remove some code that is basically a no-op. Code above this already ensures that the buffer is terminated. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@155515 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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https://origsvn.digium.com/svn/asterisk/trunk ........ r155467 | mmichelson | 2008-11-07 17:41:44 -0600 (Fri, 07 Nov 2008) | 12 lines Set the invite state to INV_CANCELLED in a place that makes more sense. Where it was set before, it was impossible to actually delay sending a CANCEL if we had not yet received a provisional response to an INVITE. (closes issue #13626) Reported by: atis Patches: 13626.patch uploaded by putnopvut (license 60) Tested by: atis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@155469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 155399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r155399 | tilghman | 2008-11-07 16:28:58 -0600 (Fri, 07 Nov 2008) | 14 lines Merged revisions 155398 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155398 | tilghman | 2008-11-07 16:27:32 -0600 (Fri, 07 Nov 2008) | 7 lines Clarify error message. (closes issue #13809) Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded by Corydon76 (license 14) Tested by: denke ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@155400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 155264 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r155264 | russell | 2008-11-07 09:42:04 -0600 (Fri, 07 Nov 2008) | 3 lines Remove a bogus ast_free() that Kevin noticed. This was probably just left over from pre-astobj2ified chan_sip. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@155272 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 155241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r155241 | russell | 2008-11-07 08:50:30 -0600 (Fri, 07 Nov 2008) | 4 lines Fix some code in chan_sip that was intended to unlink multiple objects from a container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would only remove a single object. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@155242 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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import gcc 4.3.2 warning fixes from trunk, with a few changes specific to this branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@153710 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 153362 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r153362 | russell | 2008-11-01 15:41:38 -0500 (Sat, 01 Nov 2008) | 3 lines Ensure that the sip_pvt properly has its refcount incremented when the scheduler holds a reference to it for session timer processing. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@153363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Merged revisions 153122 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r153122 | tilghman | 2008-10-31 11:35:21 -0500 (Fri, 31 Oct 2008) | 10 lines Merged revisions 153114 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r153114 | tilghman | 2008-10-31 11:30:32 -0500 (Fri, 31 Oct 2008) | 3 lines Turn off qualify on uncached realtime peers. (Closes issue #13383) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@153123 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 152920 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r152920 | russell | 2008-10-30 14:40:15 -0500 (Thu, 30 Oct 2008) | 3 lines Fix the sip_peer reference count with respect to scheduler entries for scheduling peer pokes, and scheduling peer poke expirations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152921 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 152915 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r152915 | russell | 2008-10-30 14:35:36 -0500 (Thu, 30 Oct 2008) | 3 lines Fix the sip_peer reference count with respect to scheduler entries for registration expirations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152919 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 152877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r152877 | russell | 2008-10-30 14:21:53 -0500 (Thu, 30 Oct 2008) | 9 lines Modify the documentation of the sip_registry struct - Remove a comment that says that the monitor thread is the only one that ever touches these objects. This is no longer the case with TCP. Also, I would eventually like to get the scheduler in its own thread, so this is just a poor assumption to make. - Note that reference counting of these objects with respect to scheduler entries is not complete. There are some leaked references when deleting scheduler entries. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152878 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 152569 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r152569 | russell | 2008-10-29 00:34:26 -0500 (Wed, 29 Oct 2008) | 15 lines Merged revisions 152539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r152539 | russell | 2008-10-29 00:23:51 -0500 (Wed, 29 Oct 2008) | 7 lines Fix an incorrect usage of sizeof() (closes issue #13795) Reported by: andrew53 Patches: chan_sip_sizeof.patch uploaded by andrew53 (license 519) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 151512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r151512 | mmichelson | 2008-10-22 10:08:25 -0500 (Wed, 22 Oct 2008) | 8 lines The logic of a strncasecmp call was reversed. (closes issue #13706) Reported by: andrew53 Patches: sip_notify_from_rfc3265.patch uploaded by andrew53 (license 519) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@151514 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 151464 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@151513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 151439 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r151439 | mmichelson | 2008-10-21 18:31:17 -0500 (Tue, 21 Oct 2008) | 3 lines Get this compiling in dev-mode ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@151441 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 151428 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r151428 | mmichelson | 2008-10-21 18:27:45 -0500 (Tue, 21 Oct 2008) | 14 lines If a peer uses any transport other than UDP, then MWI will fail for that peer since sip_alloc will allocate a sip_pvt with a default transport of UDP. This change resets the socket type immediately after allocating the sip_pvt in sip_send_mwi_from_peer, so that the proceeding call to create_addr_from_peer does not fail right away. The socket data from the peer is properly copied to the sip_pvt in create_addr_from_peer. (closes issue #13710) Reported by: andrew53 Patches: sip_notify_use_tcp.patch uploaded by andrew53 (license 519) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@151430 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |