mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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5469 Commits (bcfe1ecaed2bf6441fcee2bc4e1edb97ac86e631)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Mark Michelson
|
9145d4bbc8 |
Merge revision 187488 to 1.6.0.
A note to all of you. Don't block revisions in a branch if you actually meant to merge them. Two very old revisions somehow didn't get merged into 1.6.0 and this change was dependent on those two old revisions. What should have taken 2 minutes has now wasted about 30 minutes of my time :( git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187555 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
f10c9a61b3 |
Merged revisions 141810,141868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r141810 | mmichelson | 2008-09-08 16:18:49 -0500 (Mon, 08 Sep 2008) | 22 lines Merged revisions 141809 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep 2008) | 14 lines Fix pedantic mode of chan_sip to only check the remote tag of an endpoint once a dialog has been confirmed. Up until that point, it is possible and legal for the far-end to send provisional responses with a different To: tag each time. With this patch applied, these provisional messages will not cause a matching problem. (closes issue #11536) Reported by: ibc Patches: 11536v2.patch uploaded by putnopvut (license 60) ........ ................ r141868 | mmichelson | 2008-09-08 17:14:40 -0500 (Mon, 08 Sep 2008) | 4 lines Um, apparently I didn't actually finish merging before committing. Bad bad bad ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Tilghman Lesher
|
92b4eb8d40 |
Merged revisions 187363 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) | 10 lines Merged revisions 187362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) | 3 lines Permit zero-length text messages in SIP. (Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal") ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@187364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
f30f904099 |
Merged revisions 186837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 | mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 lines Fix bad merge from fix for issue 13867. (closes issue #14686) Reported by: davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Mark Michelson
|
0af241cd79 |
Remove an invalid call to free memory.
A bad merge from trunk to 1.6.0 meant freeing memory that should not be freed. In trunk, pkt->data is an ast_str, but in 1.6.0, it is allocated in the same chunk of memory as the sip_pkt. This only affects 1.6.0. (closes issue #14819) Reported by: cwolff09 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Kevin P. Fleming
|
01cbad2bd6 |
Merged revisions 186461 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
|
04670fdac8 |
Merged revisions 186101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186105 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Tilghman Lesher
|
5225fc08dc |
Merged revisions 186060 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@186061 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
|
ee5f6af8c6 |
Merged revisions 164602 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r164602 | russell | 2008-12-16 08:17:45 -0600 (Tue, 16 Dec 2008) | 7 lines Fix usage of the DAHDI_VMWI ioctl. (closes issue #14090) Reported by: alecdavis Patches: chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license 585) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185960 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
Kevin P. Fleming
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901b525432 |
Merged revisions 185953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior. this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
David Vossel
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5e9150506c |
Merged revisions 185846 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) | 16 lines Merged revisions 185845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491 Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno. Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries. Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct. When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite. In this case, it is in response to the glare invite and must be dealt with or the call is dropped. I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261. Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well. (closes issue #12013) Reported by: alx Review: http://reviewboard.digium.com/r/213/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185847 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
David Brooks
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bf746fcb47 |
Merged revisions 185363 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500 (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185426 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Richard Mudgett
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ae65af4244 |
Merged revisions 185123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185125 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Richard Mudgett
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29aa5c71a5 |
Merged revisions 185122 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines Make chan_misdn BRI TE side normally defer channel selection to the NT side. Channel allocation collisions are not handled by chan_misdn very well. This patch simply avoids the problem for BRI only. For PRI, allocation collisions are still possible but less likely since there are simply more channels available and each end could use a different allocation strategy. misdn.conf options available: te_choose_channel - Use to force the TE side to allocate channels. method - Specify the channel allocation strategy. (closes issue #13488) Reported by: Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes, festr ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@185124 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Joshua Colp
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1c3ca72745 |
Merged revisions 184948 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines Merged revisions 184947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines Improve our handling of T38 in the initial INVITE from a device. We now answer with matching media streams to what is requested. If an INVITE is received with both a T38 and RTP media stream this means we answer with both. For any outgoing calls created as a result of this inbound one no T38 is requested in the initial INVITE. Instead if we start receiving udptl packets we trigger a reinvite on the outbound side. (closes issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu Review: http://reviewboard.digium.com/r/208/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184949 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Russell Bryant
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87be216c37 |
Merged revisions 184910 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 Mar 2009) | 4 lines Fix build error when chan_h323 is not being built. (reported by cai1982 in #asterisk-dev) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184911 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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f505cb1c08 |
Merged revisions 184838 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 | russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines Simplify chan_h323 build to not require a second run of "make". (closes issue #14715) Reported by: jthurman Patches: h323-makefile-1.6.0.7-rc2.patch uploaded by jthurman (license 614) h323-makefile-1.6.1.0-rc3.patch uploaded by jthurman (license 614) h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614) Tested by: tzafrir, russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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28b8ea89dd |
Merged revisions 184566 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | 16 lines Merged revisions 184565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls. If calls were placed using an IP address or hostname the global nat setting was copied over but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP actions. (closes issue #14546) Reported by: acunningham ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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b2ca3bfb1e |
Merged revisions 184280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 | file | 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) | 5 lines Fix issue with a T38 reinvite being sent even if not configured to do so. If we receive a T38 request negotiate control frame we should only attempt to do so if the option is enabled on the dialog. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184281 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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668adab048 |
Merged revisions 184037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184037 | russell | 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) | 6 lines Exclude slin16, siren7, and siren14 from bandwidth=low and =medium The default codec configuration for chan_iax2 is bandwidth=low. I noticed slin16 being negotiated as the codec in some test calls, but that no longer happens after this change. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@184038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Leif Madsen
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f9d542aab8 |
Merged revisions 183701 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r183701 | lmadsen | 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009) | 7 lines Fixes a documentation error introduced during the CLI cleanup at AstriDevCon 2008. (closes issue #14655) Reported by: ulogic Patches: chan_dahdi.patch uploaded by ulogic (license 728) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183702 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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b95912e533 |
Merged revisions 183560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183560 | russell | 2009-03-20 12:00:58 -0500 (Fri, 20 Mar 2009) | 10 lines Merged revisions 183559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183559 | russell | 2009-03-20 11:53:25 -0500 (Fri, 20 Mar 2009) | 2 lines Fix a crash in IAX2 registration handling found during load testing with dvossel. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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84ab8b9963 |
Merged revisions 183321 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183321 | tilghman | 2009-03-19 14:17:31 -0500 (Thu, 19 Mar 2009) | 15 lines Merged revisions 183319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183319 | tilghman | 2009-03-19 14:15:33 -0500 (Thu, 19 Mar 2009) | 8 lines Delay signalling progress until a PRI channel really signals progress. (closes issue #13034) Reported by: klaus3000 Patches: 20090316__bug13034.diff.txt uploaded by tilghman (license 14) patch_trunk_183progress_klaus3000.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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2099b522d5 |
Merged revisions 183117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r183117 | mmichelson | 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) | 20 lines Merged revisions 183115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183115 | mmichelson | 2009-03-19 11:04:02 -0500 (Thu, 19 Mar 2009) | 14 lines Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use." A user was having an issue where if an outgoing SIP call was canceled, the SIP device would remain in use if we had not received any response to the initial INVITE we sent out. The SIP device would remain in use until the autocongestion timer was exhausted. I tracked down the cause of this to be the section of code I am removing here. I asked several people what the purpose of this code was meant to be, but no one could give me any sort of answer as to why this was here. The person who was having this issue has been using this patch for several months and it has stopped the problems they have had. AST-196 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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369b1b702a |
Merged revisions 183108 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jeff Peeler
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03bf1455dd |
Merged revisions 183028 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@183029 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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e047ec4d72 |
Merged revisions 182847 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jeff Peeler
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3e6a72ae73 |
Merged revisions 182722 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182723 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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a8cf0b048b |
Merged revisions 182282 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182283 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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7faf648dab |
Merged revisions 182211 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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ee6dcca4f2 |
Merged revisions 182022 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@182036 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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504ae23462 |
Merged revisions 181769 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181770 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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000375fcce |
Merged revisions 181371 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181372 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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ed9ddfd8a5 |
Merged revisions 181345 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181352 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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be3fc819ab |
Merged revisions 181296 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181297 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jeff Peeler
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95c0766715 |
add missing header file
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181284 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jeff Peeler
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368b57494b |
Merged revisions 181135 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181137 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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d90ef47592 |
Merged revisions 181032-181033 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@181034 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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c6837adabc |
If a port is specified when dialing a peer then use it.
(closes issue #14626) Reported by: acunningham git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180799 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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b332b085ee |
Ensure that the new outgoing dialog to a peer is able to set the socket details, even if the default is present.
(closes issue #14480) Reported by: jon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@180718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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f805a657a3 |
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/1.6.0@179473 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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8c5803b286 |
Merged revisions 179219 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@179220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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73af4eb39c |
Merged revisions 178871 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178874 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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e515340cc2 |
Merged revisions 178767 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178768 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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be2caea0fa |
Merged revisions 178213 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@178224 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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David Vossel
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e938398385 |
Fixes issue with undefined audio codecs in chan_iax2
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jeff Peeler
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5b4fb69323 |
Remove invalid ast_free calls for static character arrays
(issue #14478) Reported by: erik_dedecker git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177445 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jeff Peeler
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9fdccc9760 |
Merged revisions 177162 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177163 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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b2a0fc1723 |
Merged revisions 177005 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@177006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Dwayne M. Hubbard
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24ef2a922e |
Merged revisions 176705 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@176709 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |