5940 Commits (8b40aa0287e789e529e0ca8ed1eb56525931a7e0)

Author	SHA1	Message	Date
Mark Michelson	64c6397bd0	Merged revisions 194484 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines Fix a race condition where a reinvite could trigger a 482 response. The loop detection/spiral detection code in chan_sip used the owner channel's state as a criterion for determining if the incoming INVITE is a looped request. The problem with this is that the INVITE-handling code happens in a different thread than the thread that marks the owner channel as being up. As a result, if a reinvite were to come in very quickly, say from another Asterisk on the same LAN, it was possible for the reinvite to arrive before the owner channel had been set to the up state. This patch corrects the problem by using the invitestate of the sip_pvt instead, since that can be guaranteed to be set correctly by the time the reinvite arrives. Since there is a switch statement further in the INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate of the sip_pvt in case we should actually be treating the channel as if it were up already. (closes issue #12215) Reported by: jpyle Patches: 12215_confirmed.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	7872538b83	Add outgoing_colp misdn.conf port parameter. ... Select what to do with outgoing COLP information on this port. 0 - Send out COLP information unaltered. (default) 1 - Force COLP to restricted on all outgoing COLP information. 2 - Do not send COLP information. outgoing_colp=0 Also fixed sending the EctInform message so it always has the required redirectionNumber parameter when the status is active. JIRA ABE-1853 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	56275abb32	Update spiral support in trunk and 1.6.X to match what is in 1.4. ... In 1.4, a SIP spiral is treated the same way as a call forward. This works much better than what is currently in trunk and 1.6.X. The code in trunk and 1.6.X did not create a new call to the recipient of the spiral, instead trying to continue the same call. In addition to just being plain wrong, this also had the side effect of only being able to spiral calls to other SIP channels. With this in place, as long as call forwards are honored, SIP spirals will work properly. This means that it will work for outbound calls made by the Queue, Dial, and Page applications. For originated calls and spool calls, however, the spiral will not work properly until a generic call forward mechanism is introduced into Asterisk. (relates to issue #13630) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193954 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	1c988d8996	add 'const' qualifiers in various places where they should have been ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	3b01ff719c	Merged revisions 193613 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines Sent wrong message to clear a call we started if the other end has not responed yet. In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet), it is not allowed to clear the call with RELEASE_COMPLETE. It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193614 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	fbad7a508d	TCP not matching valid peer. ... find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it. Review: http://reviewboard.digium.com/r/236/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	86d63dc261	Merged revisions 193262 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) | 9 lines "misdn show config" segfaults asterisk, if no MSN lists (closes issue #14976) Reported by: alecdavis Patches: misdn_config.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, FabienToune ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	90f76fcfba	Merged revisions 193050 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines Give a more helpful message when an incoming call's dialed extension does not match. Added the dialed extension and context to the chan_misdn messages warning that the dialed number cannot be matched in the dialplan. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e4149506a1	Send DTMF frame before playing back audio. ... (closes issue #14858) Reported by: barryf Patches: 20090507__bug14858.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	01e5a86e1a	Merged revisions 192932 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines Eliminate repetition of fullcontact during reconstruction. If the fullcontact field appears in both the sippeers and the sipregs table, then during reconstruction of the field, it will otherwise be doubled. (closes issue #14754) Reported by: Alexei Gradinari Patches: 20090506__bug14754.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	9936f0ca14	Fix a bug where a timer would be created but not acknowledged. ... This scenario crept up if chan_iax2 was loaded with no configuration file present. It would create a timer and tell it to go at an interval but the thread that normally acknowledges it would not be created because no configuration file was present. The timer will now be closed if no configuration file is present. (closes issue #15014) Reported by: madkins git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	19916d118d	Merged revisions 192633 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled. (closes issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded by dimas (license 88) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	7019ff68db	Fixed crashes from issue8824 review board channel locking changes. ... The local struct ast_party_connected_line connected_caller variable was uninitialized when the copy function was called. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Fredrickson	965b0f328e	Revert CPC patch for now, until I decide whether or not it all should be merged into libss7/1.0 (It's still in the bug13495 branch and in libss7/trunk) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	2e7c1e3613	Fix a bug with setting t38pt_udptl at the user or peer level. ... If an incoming call authenticated as a user or peer and t38pt_udptl was not set to yes in general then no UDPTL session would be present and any T38 related things would fail. This commit changes it so that if after authenticating T38 is enabled but no UDPTL session is present one will be created. (issue AST-215) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	0d44a84870	Merged revisions 192213 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines global mohinterpret setting is ignored mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers. (closes issue #14728) Reported by: dimas Patches: v1-14728.patch uploaded by dimas (license 88) Tested by: dimas, dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	a0766f8113	Conditional include ioctl's to change EC policy based on DAHDI caps. ... This feels like a sane change (wouldn't compile without this addition), but I'm not intimately familiar with this code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191739 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a3229fd3e2	Merged revisions 191559 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1. (closes issue #14993) Reported by: BigJimmy Patches: causepatch uploaded by BigJimmy (license 371) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	451c59aa18	Set debug message back to DEBUG level. ... (closes issue #15007) Reported by: hulber git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	d9d2779008	Add buffer and echo canceller control to CHANNEL() dialplan function for DAHDI channels ... Adds ability for CHANNEL() dialplan function, when used on DAHDI channels, to temporarily change the number of buffers and/or the buffer policy, and also to enable, disable, or switch the echo canceller between FAX/data and voice modes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b6d2a4a7a8	Make H.323 compile with FDLEAK detection code enabled ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	d35fd35ae3	Outgoing PTP redirected calls did not wait for the COLR from the redirected-to party. ... For outgoing PTP redirected calls, you now need to use the inhibit(i) option on all of the REDIRECTING statements before dialing the redirected-to party. You still have to set the REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The PTP call will update the redirecting-to presentation when it becomes available and queue the redirecting update to the calling channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ca138fc807	Consistent SSL/TLS options across conf files ... ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though. Review: http://reviewboard.digium.com/r/237/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Fredrickson	a082ad616f	Add support setting CPC from channel variable ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Fredrickson	6389ec15fb	Make sure that we do not clear the down flag on the BRI during PTMP link transients ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	89d06c7759	Make PTP DivertingLegInformation3 message behavior closer to the specifications. ... * Wait for a DivertingLegInformation3 message after receiving a DivertingLegInformation1 message to complete the redirecting-to information before queuing a redirecting update to the other channel. * A DivertingLegInformation2 message should be responded to with a DivertingLegInformation3 when the COLR is determined. If the call could or does experience another redirection, you should manually determine the COLR to send to the switch by setting REDIRECTING(to-pres) to the COLR and setting REDIRECTING(to-num) = ${EXTEN}. * A DivertingLegInformation2 message must have an original called number if the redirection count is greater than one. Since Asterisk does not keep track of this information, we can only indicate that the number is not available due to interworking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	8f0b88c8c8	TLS/SSL private key option ... Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up. Review: http://reviewboard.digium.com/r/234/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	c95c065903	There is no need to use the struct ast_party_connected_line.source update values. ... The messages sent by a technology when a connected line update is received are best determined by the current call state of the channel. The struct ast_party_connected_line.source value is really only useful as a possible tracing aid. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c878b21887	Fix a build error. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	cba19c8a67	Convert the ast_channel data structure over to the astobj2 framework. ... There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	f314c5f13f	Fix nat setting on RTP instances. ... (closes issue #14827) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	bbcf30787d	Merged revisions 190356 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190356 | russell | 2009-04-23 16:07:07 -0500 (Thu, 23 Apr 2009) | 2 lines Remove a bogus ast_channel_unlock(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	80b8d34377	Merged revisions 190286 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6 lines Fix a bug in chan_local glare hangup detection. If both sides of a Local channel were hung up at around the same time it was possible for one thread to destroy the local private structure and have the other thread immediately try to remove the already freed structure from the local channel list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	2f8da7106d	Make chan_h323 respect packetization settings and fix small reload issue. ... Previously, packetization settings were ignored and now they are not. A new config option 'autoframing' has been added to mirror the way chan_sip handles it. Turning on the autoframing option (available both as a global option or per peer) overrides the local settings with the remote packetization settings. Testing was performed with varying packetization levels with the following codecs: ulaw, alaw, gsm, and g729. Also, an unrelated config reload issue has been fixed in the case of the config file not changing. (closes issue #12415) Reported by: pj Patches: 2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7), modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9cbd90a864	Do not continue to receive DTMF, when the channel is hungup and about to be destroyed. ... (closes issue #14858) Reported by: barryf Patches: 20090421__bug14858.diff.txt uploaded by tilghman (license 14) Tested by: barryf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	00d7c4fefc	Fixes segfault when switching UDP to TCP in sip.conf after reload. ... If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload. The problem is the socket type is changed to TCP but the fd may still be present for UDP. Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present. Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found. (closes issue #14727) Reported by: pj Tested by: dvossel Review: http://reviewboard.digium.com/r/229/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	6bb2b6c096	Added CCBS/CCNR Party A support and enhanced COLP support. ... This change adds the following features to chan_misdn: * CCBS/CCNR Party A support for PTMP and PTP modes. * Enhances COLP support for call diversion and explicit call transfer. These enhanced features require a modified version of mISDN. The latest modified mISDN v1.1.x based version is available at: http://svn.digium.com/svn/thirdparty/mISDN/trunk http://svn.digium.com/svn/thirdparty/mISDNuser/trunk Taged versions of the modified mISDN code are available under: http://svn.digium.com/svn/thirdparty/mISDN/tags http://svn.digium.com/svn/thirdparty/mISDNuser/tags Review: http://reviewboard.digium.com/r/218/ Merged from team/rmudgett/misdn_facility branch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Doug Bailey	9c2ff7bb1e	Merged revisions 189391 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189391 | dbailey | 2009-04-20 14:10:56 -0500 (Mon, 20 Apr 2009) | 4 lines Clean up problem with manager implementation of mmap where it was not testing against MAP_FAILED response. Got rid of shadowed variable used in processign the mmap results. Change test of mmap results to compare against MAP_FAILED ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	b5b18b7810	Fix a bug with non-UDP connections that caused dialogs to not get freed. ... This issue crept up because of a reference count issue on non-UDP based dialogs. The dialog reference count was increased when transmitting a packet reliably but never decreased. This caused the dialog structure to hang around despite being unlinked from the dialogs container. (closes issue #14919) Reported by: vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	437eec8423	Merged revisions 189203 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening. (closes issue #14091) Reported by: evandro Patches: autologoff.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/225/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	13af9ed74c	Merged revisions 188833,189134 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@189137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	4bf5e1b805	Prevent a crash when SIP blonde transferring an unbridged call. ... 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: https://origsvn.digium.com/svn/asterisk/trunk@189097 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	6d2b446384	Merged revisions 188946 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@188947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	b0347267b3	Merged revisions 188937 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@188938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ecc3c7c4a6	Merged revisions 188835 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@188836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a9895c6833	SIP state notify reorganization ... 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: https://origsvn.digium.com/svn/asterisk/trunk@188742 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	ac48378d28	Fix a bug with the dahdi_setoption callback in chan_dahdi. ... 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: https://origsvn.digium.com/svn/asterisk/trunk@188705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	b2e77d5f5d	Merged revisions 188646 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@188647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	2fd695d23c	Don't try to do anything in pri_check_restart with service messages unless ... libpri supports it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188515 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	50ecc19ca0	change some capitalization ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago