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2009-04-17Merged revisions 188833,189134 via svnmerge from rmudgett2-3/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled. Leave jitter setting alone. JIRA ABE-1835 ........ r189134 | rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines Modifed/added some debug messages. JIRA ABE-1835 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@189137 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-17Prevent a crash when SIP blonde transferring an unbridged call.mmichelson1-5/+1
If one attempts to use the attended transfer button on a SIP phone to transfer an unbridged call (such as a call to an IVR) but hangs up while the target of the transfer is still ringing, we need to not crash. The problem was that ast_hangup was called from outside the channel thread. AST-211 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@189097 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-17Merged revisions 188946 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines Fix a bug where a value used to create the channel name was bogus. This commit fixes the scenario where an incoming call is authenticated using a peer entry. Previously the channel name was created using either the username setting from the sip.conf entry or the IP address that the call came from. Now the channel name will be created using the peer name itself. This commit will not change the way the channel name is generated for users or friends. (closes issue #14256) Reported by: Nick_Lewis Patches: chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, file ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188947 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-17Merged revisions 188937 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4 lines Fix a situation where the DAHDI channel private structure lock was not unlocked when it should have been. (issue AST-210) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188938 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-16Merged revisions 188835 via svnmerge from tilghman1-5/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) | 7 lines Only update realtime, if global option rtupdate != false (closes issue #14885) Reported by: deepesh Patches: 20090413__bug14885.diff.txt uploaded by tilghman (license 14) Tested by: deepesh ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188836 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-16SIP state notify reorganizationdvossel1-88/+119
What I've done here is simply break up how a state NOTIFY is built. Originally both the XML and sip header information were built within the same function. While this does work, it does not allow for the creation of multipart/related message bodies that can contain multiple XML entries with only one sip header. Now a separate function builds the XML for each notify. This patch also makes maintaining and modifying state notifications in the future much less of a pain. Review: http://reviewboard.digium.com/r/224/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188742 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-16Fix a bug with the dahdi_setoption callback in chan_dahdi.file1-0/+2
This function incorrectly reported success even if the option was unsupported. This was exposed by the options to change the underlying channel format. The function now returns a failure if the option is unsupported. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188705 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-15Merged revisions 188646 via svnmerge from dvossel1-0/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines National prefix inserted even when caller ID not available When the caller ID is restricted, the expected behavior is for the caller id to be blank. In chan_dahdi, the national prefix is placed onto the callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank. (closes issue #13207) Reported by: shawkris Patches: national_prefix.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/220/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188647 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-15Don't try to do anything in pri_check_restart with service messages unlessjpeeler1-0/+4
libpri supports it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188515 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-14change some capitalizationjpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188378 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-14Add service maintenance message supportjpeeler1-18/+350
This is the companion commit to libpri r732. Service messages are now supported for switch types 4ess/5ess. A new option service_message_support has been added to chan_dahdi.conf and is noted in the sample config file. The service message support is turned off by default. The current implementation relies on AstDB to keep track of channel state, which allows the statuses to be preserved across Asterisk restarts. Below is a description of the storage format. The state and reason for the service state are in the form <state>:<reason>, where: <state> ::= { 'O' } // 'O' – Out Of Service <reason> ::= { '0' | '1' | '2' | '3' }, where: '0' – No reason (backwards compatibility) '1' – NEAR END '2' – FAR END '3' – both NEAR and FAR END The new CLI commands to handle channel service state are: pri service disable channel <chan> pri service enable channel <chan> Many people contributed to the development of this functionality. Because I entered at the very end I do not know the exact history. Special thanks to all who moved the bug forward one way or another: cmaj, PCadach, markster, mattf, drmac, MikeJ, serge-v, murf, kanelbullar, Seb7, tilghman, lmadsen, and especially dhubbard (he answered lots of my questions and did a large portion of the work) (closes issue #3450) Reported by: cmaj git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188342 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-14Fix a bug with the change I made yesterday to outbound proxy support.file1-5/+1
Per discussion with oej on IRC we need the actual IP address, not the outbound proxy IP address, in the sa field. This change matches the already existing code for all other uses of the outbound proxy setting. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188247 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-13Fix a bug where using an outbound proxy would cause the local address to be ↵file1-0/+1
127.0.0.1. Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will be sending to. This has to be done because the logic that determines what local IP address to use in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address we are sending to. (closes issue #12006) Reported by: mnicholson git-svn-id: http://svn.digium.com/svn/asterisk/trunk@188067 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-10Fix module embedding for chan_h323.jpeeler1-6/+7
Include libchanh323.a in the modules.link file so that all the symbols can be resolved at link time. (closes issue #11966) Reported by: dome Patches: issue_11966.patch uploaded by kpfleming (license 421) Tested by: jpeeler git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187906 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-10Indicating connected line or redirecting updates were missing a call to lock ↵mmichelson1-0/+1
the local_pvt. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187830 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-10Fix some uninitialized memory notices that appeared under valgrind.file7-21/+21
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187772 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-10ast_strdup failures aren't really failures if the original value was NULL.mmichelson1-4/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187714 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-10Ensure pvt is not NULL before dereferencing it.tilghman1-2/+2
(closes issue #14784) Reported by: pj git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187674 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-10Miscellaneous minor changes to chan_misdn.rmudgett3-207/+245
* Miscellaneous spacing and comment changes. * Minor code rearangements. * Miscellaneous doxygen comments. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187635 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-10Make chan_misdn_log() avoid generating the log message if logging is disabled.rmudgett1-17/+27
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187634 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-09Add a new option, mwi_from, to sip.conf.mmichelson1-0/+26
This allows for you to change the From header for outgoing MWI NOTIFY requests. Prior to this, the best you could do was to set a callerid in the general section of sip.conf. The problem was that this was used for all outbound requests, not just MWI NOTIFY requests. AST-201 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187560 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-09Merged revisions 187484 via svnmerge from mmichelson1-6/+19
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187484 | mmichelson | 2009-04-09 13:51:20 -0500 (Thu, 09 Apr 2009) | 18 lines Handle a SIP race condition (reinvite before an ACK) properly. RFC 5047 explains the proper course of action to take if a reINVITE is received before the ACK from a previous invite transaction. What we are to do is to treat the reINVITE as if it were both an ACK and a reINVITE and process it normally. Later, when we receive the ACK we had been expecting, we will ignore it since its CSeq is less than the current iseqno of the sip_pvt representing this dialog. (closes issue #13849) Reported by: klaus3000 Patches: 13849_v2.patch uploaded by mmichelson (license 60) Tested by: mmichelson, klaus3000 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187488 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-09Allow '/' in username portion of register; this is a regression.tilghman1-1/+1
(closes issue #14668) Reported by: Netview git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187381 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-09Merged revisions 187362 via svnmerge from tilghman1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@187363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-09Do not try to send the format read/format write/make compatible options over ↵file1-0/+4
IAX2. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187361 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-09Add support for allowing the channel driver to handle transcoding.file1-0/+22
This was accomplished using a set of options and the setoption channel callback. The core calls into the channel driver using these options and the channel driver either returns success or failure. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187360 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Remove duplicate prototype for temp_peer().russell1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187105 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Update some comments and resolve potential memory corruption in chan_sip.russell1-28/+13
While browsing chan_sip the other day, I noticed this dangerous code in dialog_needdestroy(). This function is an ao2_callback. It is absolutely _not_ okay to unlock the container from within this function. It's also not clear why it was useful. Given that it could cause memory corruption, I have removed it. There was also a TODO comment left describing a potential implementation of an improvement to the needdestroy handling. I'm not convinced that what was described is the best choice here, so I have briefly described the way that this function is used today that could be improved. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186928 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Add lastms to the require API call.tilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186899 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Fix bad merge from fix for issue 13867.mmichelson1-3/+1
(closes issue #14686) Reported by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186837 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-06Fix problem when authenticating a non-RTP dialog.file1-2/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186653 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-06Add support for changing the outbound codec on a SIP call usingfile1-2/+10
a dialplan variable. This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls the codec offered for an outgoing SIP call. This is much like the SIP_CODEC dialplan variable and has the same restrictions. The codec set must be one that is configured for the call. (closes issue #13243) Reported by: samdell3 Patches: 13243.diff uploaded by file (license 11) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186624 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03This commit introduces COLP/CONP and Redirecting party information into ↵mmichelson17-1074/+2813
Asterisk. The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186525 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Merged revisions 186458 via svnmerge from kpfleming1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@186461 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Add better support for relaying success or failure of the ast_transfer() API ↵file2-2/+27
call. This API call now waits for a special frame from the underlying channel driver to indicate success or failure. This allows the return value to truly convey whether the transfer worked or not. In the case of the Transfer() dialplan application this means the value of the TRANSFERSTATUS dialplan variable is actually true. (closes issue #12713) Reported by: davidw Tested by: file git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186382 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186081 via svnmerge from kpfleming1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@186101 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merge in the RTP engine API.file10-833/+805
This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186078 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186059 via svnmerge from tilghman1-8/+107
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: http://svn.digium.com/svn/asterisk/trunk@186060 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 185952 via svnmerge from kpfleming1-11/+20
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: http://svn.digium.com/svn/asterisk/trunk@185953 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-01Merged revisions 185845 via svnmerge from dvossel1-13/+30
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: http://svn.digium.com/svn/asterisk/trunk@185846 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Improve performance of the code handling the frame queue in chan_iax2.russell1-124/+96
In my tests that exercised full frame handling in chan_iax2, the version with these changes took 30% to 40% of the CPU time compared to the same test of Asterisk trunk before these modifications. While doing some profiling for <http://reviewboard.digium.com/r/205/>, one function that caught my eye was network_thread() in chan_iax2.c. After the things that I was working on there, it was the next target for analysis and optimization. I used oprofile's source annotation functionality and found that the loop traversing the frame queue in network_thread() was to blame for the excessive CPU cycle consumption. The frame_queue in chan_iax2 previously held all frames that either were pending transmission or had been transmitted and are still pending acknowledgment. In network_thread(), the previous code would go back through the main for loop after reading a single incoming frame or after being signaled because a frame had been queued up for initial transmission. In each iteration of the loop, it traverses the entire frame queue looking for frames that need to be transmitted. On a busy server, this could easily be quite a few entries. This patch is actually quite simple. The frame_queue has become only a list of frames pending acknowledgment. Frames that need to be transmitted are queued up to a dedicated transmit thread via the taskprocessor API. As a result, the code in network_thread() becomes much simpler, as its only job is to read incoming frames. In addition to the previously described changes, this patch includes some additional changes to the frame_queue. Instead of one big frame_queue, now there is a list per call number to further reduce wasted list traversals. The biggest impact of this change is in socket_process(). For additional details on testing and test results, see the review request. Review: http://reviewboard.digium.com/r/212/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185432 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Merged revisions 185362 via svnmerge from dbrooks1-7/+7
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: http://svn.digium.com/svn/asterisk/trunk@185363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 185121 via svnmerge from rmudgett1-4/+5
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: http://svn.digium.com/svn/asterisk/trunk@185123 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 185120 via svnmerge from rmudgett1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@185122 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 184947 via svnmerge from file1-268/+235
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: http://svn.digium.com/svn/asterisk/trunk@184948 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Fix build error when chan_h323 is not being built.russell1-0/+2
(reported by cai1982 in #asterisk-dev) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184910 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-29Simplify chan_h323 build to not require a second run of "make".russell1-15/+10
(closes issue #14715) Reported by: jthurman Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614) Tested by: tzafrir, russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184838 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Improve timing interface to remember which provider provided a timerkpfleming1-12/+11
The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error. This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider. (closes issue #14697) Reported by: moy Review: http://reviewboard.digium.com/r/211/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@184762 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Merged revisions 184565 via svnmerge from file1-0/+2
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/trunk@184566 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-25Improve performance of the ast_event cache functionality.russell5-15/+2
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/trunk@184339 f38db490-d61c-443f-a65b-d21fe96a405b
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