asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	84fd750c10	Fix a bug with the change I made yesterday to outbound proxy support. 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: https://origsvn.digium.com/svn/asterisk/trunk@188247 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	75dba8ca1d	Fix a bug where using an outbound proxy would cause the local address to be 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: https://origsvn.digium.com/svn/asterisk/trunk@188067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	a8ffff75bb	Fix module embedding for chan_h323. 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: https://origsvn.digium.com/svn/asterisk/trunk@187906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	86d6af95ef	Indicating connected line or redirecting updates were missing a call to lock the local_pvt. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	8e4b5df187	Fix some uninitialized memory notices that appeared under valgrind. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	9b580ea645	ast_strdup failures aren't really failures if the original value was NULL. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	15e040d3f3	Ensure pvt is not NULL before dereferencing it. (closes issue #14784) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	b0cfe6e1f2	Miscellaneous minor changes to chan_misdn. * Miscellaneous spacing and comment changes. * Minor code rearangements. * Miscellaneous doxygen comments. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	b89dce07b4	Make chan_misdn_log() avoid generating the log message if logging is disabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4d74179f20	Add a new option, mwi_from, to sip.conf. 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: https://origsvn.digium.com/svn/asterisk/trunk@187560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	e53bd994d0	Merged revisions 187484 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@187488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	7304ac444e	Allow '/' in username portion of register; this is a regression. (closes issue #14668) Reported by: Netview git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3a220874cc	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/trunk@187363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	e2a336124f	Do not try to send the format read/format write/make compatible options over IAX2. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	abcc0b9397	Add support for allowing the channel driver to handle transcoding. 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: https://origsvn.digium.com/svn/asterisk/trunk@187360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c4058865dd	Remove duplicate prototype for temp_peer(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0fab071d13	Update some comments and resolve potential memory corruption in chan_sip. 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: https://origsvn.digium.com/svn/asterisk/trunk@186928 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b289374dfe	Add lastms to the require API call. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186899 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	21dd185512	Fix bad merge from fix for issue 13867. (closes issue #14686) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	369ca78928	Fix problem when authenticating a non-RTP dialog. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	4eaa651a8a	Add support for changing the outbound codec on a SIP call using 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: https://origsvn.digium.com/svn/asterisk/trunk@186624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	6f53ed4c67	This commit introduces COLP/CONP and Redirecting party information into 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: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	3525e37e63	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/trunk@186461 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	2d9c6ef3d5	Add better support for relaying success or failure of the ast_transfer() API 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: https://origsvn.digium.com/svn/asterisk/trunk@186382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	612fc2e7e3	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/trunk@186101 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	63de834395	Merge in the RTP engine API. 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: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	08971ce205	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/trunk@186060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	d99d2f22cd	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/trunk@185953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	729f225225	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/trunk@185846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8dfcd7e418	Improve performance of the code handling the frame queue in chan_iax2. 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: https://origsvn.digium.com/svn/asterisk/trunk@185432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	b90ee93f70	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/trunk@185363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	9fd753a30e	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/trunk@185123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	5e707f2ded	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/trunk@185122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	aa056be678	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/trunk@184948 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	3ec3742a97	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/trunk@184910 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	39c95555af	Simplify chan_h323 build to not require a second run of "make". (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: https://origsvn.digium.com/svn/asterisk/trunk@184838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	9381bff79d	Improve timing interface to remember which provider provided a timer 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: https://origsvn.digium.com/svn/asterisk/trunk@184762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	0580121cee	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/trunk@184566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ee77b475f2	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: https://origsvn.digium.com/svn/asterisk/trunk@184339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	9ae51a21c0	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/trunk@184280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	7460afdd46	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/trunk@184037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	da2230adf0	SIP preferred codec only feature Added an option to respond to a SIP invite with only the single most preferred joint codec. This limits the options of what codecs the other side can use. (closes issue #12485) Reported by: bamby Review: http://reviewboard.digium.com/r/206/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	9a6bf5f9c6	Removed trailing whitespace in chan_misdn files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Leif Madsen	18b4508c8e	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/trunk@183701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	f4d0347d02	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/trunk@183560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	7ed6354989	Fix chan_sip so it builds. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	830602bc69	Remove duplicate <description> inside the xml documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	2e516d0c26	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/trunk@183321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	abb71e3d55	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/trunk@183117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	12bd92898d	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/trunk@183108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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5889 Commits (34080f9cbecd9094088821f1b287742103896079)