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2009-03-27Merged revisions 184566 via svnmerge from file1-0/+2
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@184612 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-25Merged revisions 184339 via svnmerge from russell5-15/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) | 35 lines Improve performance of the ast_event cache functionality. This code comes from svn/asterisk/team/russell/event_performance/. Here is a summary of the changes that have been made, in order of both invasiveness and performance impact, from smallest to largest. 1) Asterisk 1.6.1 introduces some additional logic to be able to handle distributed device state. This functionality comes at a cost. One relatively minor change in this patch is that the extra processing required for distributed device state is now completely bypassed if it's not needed. 2) One of the things that I noticed when profiling this code was that a _lot_ of time was spent doing string comparisons. I changed the way strings are represented in an event to include a hash value at the front. So, before doing a string comparison, we do an integer comparison on the hash. 3) Finally, the code that handles the event cache has been re-written. I tried to do this in a such a way that it had minimal impact on the API. I did have to change one API call, though - ast_event_queue_and_cache(). However, the way it works now is nicer, IMO. Each type of event that can be cached (MWI, device state) has its own hash table and rules for hashing and comparing objects. This by far made the biggest impact on performance. For additional details regarding this code and how it was tested, please see the review request. (closes issue #14738) Reported by: russell Review: http://reviewboard.digium.com/r/205/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@184343 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-25Merged revisions 184280 via svnmerge from file1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@184283 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-24Merged revisions 184037 via svnmerge from russell1-0/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@184041 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-23Merged revisions 183701 via svnmerge from lmadsen1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@183704 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-20Merged revisions 183560 via svnmerge from russell1-5/+10
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@183564 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-20Merged revisions 183511 via svnmerge from eliel1-3/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r183511 | eliel | 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) | 2 lines Remove duplicate <description> inside the xml documentation. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@183519 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183321 via svnmerge from tilghman1-0/+9
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@183337 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183117 via svnmerge from mmichelson1-5/+0
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@183122 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-19Merged revisions 183108 via svnmerge from file1-184/+41
https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 | file | 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) | 11 lines Improve our triggering of a T38 switchover internally when triggered by a received reinvite. Previously we reached across the channel bridge to get the other party's SIP dialog structure in order to trigger an outgoing reinvite. This is extremely dangerous to do and only works if bridged to another SIP channel. This patch changes this to use the T38 control frame method of requesting a switchover. This change also causes the SIP channel driver to propogate back whether the switchover worked or not instead of blindly accepting the incoming T38 reinvite. Review: http://reviewboard.digium.com/r/200/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@183111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Merged revisions 183028 via svnmerge from jpeeler1-1/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r183028 | jpeeler | 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) | 4 lines Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@183031 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Merged revisions 182847 via svnmerge from russell2-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@182947 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Merged revisions 182722 via svnmerge from jpeeler6-75/+126
https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 | jpeeler | 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) | 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@182725 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-17Merged revisions 182408 via svnmerge from rmudgett1-35/+36
https://origsvn.digium.com/svn/asterisk/trunk ........ r182408 | rmudgett | 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009) | 8 lines OPENR2 uses an incorrect string value if the extension delimiter is not present. * Fixed OPENR2 using an incorrect string value if the extension delimiter is not present in the Dial() function. This was fixed for SS7 and PRI in trunk -r172400. * Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others. * Removed trailing whitespace that appeared with OPENR2. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@182409 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-16Add MFC/R2 support for chan_dahdi.russell1-23/+1544
This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182355 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-16Merged revisions 182281 via svnmerge from dvossel1-1/+18
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 | dvossel | 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) | 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182282 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-16Merged revisions 182208 via svnmerge from tilghman1-11/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182208 | tilghman | 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) | 7 lines Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182211 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-13Fix an issue with requesting a T38 reinvite before the call is answered.file1-2/+6
The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182022 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-13improve a bit of suboptimal codekpfleming1-5/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181985 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-12Merged revisions 181768 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 | mmichelson | 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) | 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181769 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181340 via svnmerge from dvossel2-32/+72
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181371 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181328 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181345 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181295 via svnmerge from file1-3/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181296 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Fix malloc debug macros to work properly with h323.jpeeler2-5/+5
The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181135 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Add missing comment that quotes RFC 3891mmichelson1-1/+14
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181033 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181029,181031 via svnmerge from mmichelson1-15/+4
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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@181032 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merge phase 1 support for the new bridging architecture.file1-0/+246
This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180369 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Resolve object matching issues related to the removal of the sip_user object.russell1-62/+70
Previously, chan_sip had both sip_peer and sip_user objects in memory. A patch went in to remove sip_user to simplify the code, since everything could be done with just sip_peer. This patch resolves some regressions found that were introduced by those changes. This code comes from svn/asterisk/team/group/sip-object-matching/. Here is a list of the changes that have been made: 1) When doing a match by name with the find_peer() function, make it much easier to specify which objects should be matched by having a parameter that specifies exactly which object types should be considered. Also, update find_by_name() to handle this parameter. Finally, update all code to use the new option values. 2) When looking up an object for an outbound request by name, consider peers only. (create_addr()) 3) Only match peers on an incoming registration request. 4) When doing authentication (except for SUBSCRIBE), look up users by name, instead of all objects by name. 5) When doing authentication (except for SUBSCRIBE), after looking for a user by name, look for a peer by IP address, instead of all objects by IP address. 6) When handling the SIP qualify CLI command or manager action, look for a peer by name, instead of any object by name. 7) When handling the SIP unregister CLI command, look for a peer by name, instead of any object by name. 9) In sip_do_debug_peer(), search for a peer by name, instead of any object by name. 9) When handling the SIPPEER() dialplan function, search for a peer by name, instead of any object by name. 10) In the following session timer related functions, st_get_se(), st_get_refresher(), and st_get_mode(), when looking for an object for a given sip_pvt using pvt->peername, look for a peer by name, instead of any object by name. 11) Fix build_peer() to properly handle the case where separate type=peer and type=user entries were specified in sip.conf. (closes issue #14505) Reported by: lmadsen Review: http://reviewboard.digium.com/r/172/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180261 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Allow for "magic" pickups to work when we wish to ignore the contextmmichelson1-1/+1
When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180155 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Please prefix default values with DEFAULToej1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@179675 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Do not try to remove a registration scheduled item if the scheduler context ↵file1-1/+3
has already been destroyed. (closes issue #14580) Reported by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@179323 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-01Properly free memory and remove scheduler entries when a transmission ↵mmichelson1-2/+4
failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@179219 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Add reload support to chan_skinny.mvanbaak1-48/+209
Special thanks goes to DEA who had to redo this patch twice because we first put unload/load support in and later redid the way we configure devices and lines. (closes issue #10297) Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA (license 3) With mods by me based on feedback from wedhorn and Russell and seanbright Tested by: DEA, mvanbaak, pj Review: http://reviewboard.digium.com/r/130/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@179122 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26IAX2 prune realtime, minor tweak to last fixdvossel1-0/+1
A return statement was missing which caused unexpected cli output. issue #14479 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178871 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26IAX2 prune realtime fixdvossel1-16/+43
Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178767 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-24Allows manager command to see if IAX link is trunked and encrypted. Displays ↵dvossel1-18/+42
what kind of encryption is enabled as well. Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not. (closes issue #14427) Reported by: snuffy Patches: iax_show_trunks.diff uploaded by snuffy (license 35) 2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review: http://reviewboard.digium.com/r/173/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178300 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-24Merged revisions 178205 via svnmerge from file1-3/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178213 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-23update the new manager commands in chan_skinny to matchmvanbaak1-5/+5
chan_sip's headers. requested by oej. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178061 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-23Changes the way keyrotation is enabled by defaultdvossel1-5/+3
Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers". Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled. (closes issue #14523) Reported by: mvanbaak git-svn-id: http://svn.digium.com/svn/asterisk/trunk@178030 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-22Add a couple of manager commands to chan_skinnymvanbaak1-143/+479
Added: SKINNYdevices SKINNYshowdevice SKINNYlines SKINNYshowline (closes issue #14521) Reported by: mvanbaak Review: http://reviewboard.digium.com/r/170/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177988 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-21On update, test against the existence of sipregs.tilghman1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177944 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-21make chan_sip.c compile on OpenBSD again.mvanbaak1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177849 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-20Set sip_request ast_str data to NULL so ast_str_copy allocates space properlyjpeeler1-0/+1
in copy_request (issue #14478) Reported by: erik_dedecker git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177624 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Modify h323 to build against PTLib as well as the older PWLibjpeeler10-110/+146
Several changes in PTLib have occurred requiring build time detection. Changes accounted for include the library name change, config option change, install location change, and a boolean type change which is handled by ast_ptlib.h. Also, the sed check has been modified to properly work with autoconf >= 2.62. (closes issue #14224) Reported by: bergolth Patches: asterisk-autoconf-sed.patch uploaded by bergolth (license 661) asterisk-pwlib-v3.patch uploaded by bergolth (license 661) Tested by: jpeeler git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177162 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Fix ordering of output for a ChannelUpdate manager event.file1-1/+1
(closes issue #14497) Reported by: vinsik Patches: chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@177005 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18T38 faxdetect should jump to the 'fax' extension for incoming calls onlydhubbard1-2/+2
The previous implementation of T38 faxdetect resulted in both sides of the call jumping to a fax extension when both sides had 't38pt_udptl=yes' and 'faxdetect=yes' in sip.conf and a 'fax' extension in the current context. This revision will jump to a 'fax' extension on incoming calls only. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176869 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17create a UDPTL structure in create_addr_from_peer() if it does not already ↵dhubbard1-0/+4
exist for T38 This is required to create a UDPTL structure in create_addr_from_peer() to handle the scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but is defined the peer's context. I tested this patch by enabling t38pt_udptl in the [general] section on one system and only enabling t38pt_udptl in a peer's context on the system sending a fax. Without the patch, the sending system will fail to initiate T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure". When this patch is applied the sending side will successfully initiate T38 negotiation. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176705 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Prior to masquerade, move the group definitions to the channel performing thetilghman1-2/+3
masq, so that the group count lingers past the bridge. (closes issue #14275) Reported by: kowalma Patches: 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176642 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merge a large set of updates to the Asterisk indications API.russell3-20/+33
This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17In this version, we can combine the queries, because we support droppingtilghman1-5/+2
nonexistent columns. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176501 f38db490-d61c-443f-a65b-d21fe96a405b
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