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2009-09-14Merged revisions 218401 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218401 | jpeeler | 2009-09-14 16:47:11 -0500 (Mon, 14 Sep 2009) | 11 lines Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor. After talking to rmudgett about some of his recent iflist locking changes, it was determined that the only place that would destroy a channel without being explicitly to do so was in handle_init_event. The loop to walk the interface list has been modified to wait to destroy the channel until the dahdi_pvt of the channel to be destroyed is no longer needed. (closes issue #15378) Reported by: samy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-14Add support for multiple interface lists.Richard Mudgett
Also unlink the sig_pri_pri.pvts[] pointer in destroy_dahdi_pvt(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-13gcc 4.4: Remove a nop memset size 0 that annoys gccTzafrir Cohen
This memset doesn't write beyond the end of the buffer. (tmpbuf has size of 4). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-13get rid of mfcr2 monitor thread condition, is problematicMoises Silva
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Cleanup approach in 217804 and don't reach inside the sig_pvt.Jeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Allow do not disturb to be set on analog channels via the CLI and AMI.Jeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Make calltoken support work with realtime users and peers.Tilghman Lesher
In the course of this, I also found that the results of ast_gethostbyname were being used incorrectly in both chan_iax2 and chan_sip, so both have been fixed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Cleaned up chan_dahdi iflist handling and locking.Richard Mudgett
* Fixed walking the iflist so it is always done with the iflock locked. * Simplified iflist walking routines. * Created chan_dahdi iflist insertion and extraction routines. * Fixed duplicate_pseudo() malloc fail handling. * Fixed infinite loop in action_dahdishowchannels() when showing a single channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Miscellaneous minor changes.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Merged revisions 217806 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r217806 | dvossel | 2009-09-10 16:06:07 -0500 (Thu, 10 Sep 2009) | 22 lines IAX2 encryption regression The IAX2 Call Token security patch inadvertently broke the use of encryption due to the reorganization of code in the socket_process() function. When encryption is used, an incoming full frame must first be decrypted before the information elements can be parsed. The security release mistakenly moved IE parsing before decryption in order to process the new Call Token IE. To resolve this, decryption of full frames is once again done before looking into the frame. This involves searching for an existing callno, checking the pvt to see if encryption is turned on, and decrypting the packet before the internal fields of the full frame are accessed. (closes issue #15834) Reported by: karesmakro Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, karesmakro Review: https://reviewboard.asterisk.org/r/355/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Fix crash during attended transfer over PRI.Jeff Peeler
The owner pointers in the sig_pri_chan structure were not getting updated in dahdi_fixup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Stop caller id transmission when offhook event detected.Jeff Peeler
This fixes the problem that would occur if an analog phone was picked up while the caller id was being sent. The caller id before sent the whole spill even after pickup and is now corrected. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Don't assign UINT_MAX to an INT. Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Include ActionID in all events that are responsed to AMI Action SIPShowRegistryOlle Johansson
(closes issue #15868) Reported by: nic_bellamy Patches: manager_SIPshowregistry_actionid.patch uploaded by nic bellamy (license 299) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-10Fix available() for SS7, MFC/R2, and pseudo channels.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-09ast_log replaced for ast_verbose in MFCR2 event notificationsMoises Silva
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-09Don't report transfer success until we actually know. 1xx messages are not ↵Olle Johansson
final. Related to #12713 Patch by oej A big thank you to file for finally fixing the transfer() dialplan application. I've been waiting for years for this. Great work! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-09Not having any TLS session to write to is a serious XMIT_ERROR. Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-09Formatting and doxygen updatesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-08Fix memory leak of sig_xxx private structures.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-08Miscellaneous minor code cleanup in mkintf().Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-08Remove duplicate entry in the sig_pri_pri private pointer array.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@217236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-08caller id number emptyDavid Vossel
parse_uri was not being given the correct scheme's, as a result, uri parsing did not parse the username correctly. One of the side effects of this is an empty caller id. (closes issue #15839) Reported by: ebroad Patches: blank_cidv2.patch uploaded by ebroad (license 878) parse_uri_fix.diff uploaded by dvossel (license 671) Tested by: ebroad, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Moving another function declared in the middle of forward declarations. Olle Johansson
Please follow the structure of the source code, thanks. Chan_sip is messy enough as it is :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Move "deprecated_username" to a flag like the others - unsigned int blah:1Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07- Doxygen additionsOlle Johansson
- Remove unused string in sip_registry -- "random" - Someone added a function in the middle of all forward declarations... Weird. Moved it out of that section. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Clean up the "offered_media" codeOlle Johansson
- Add variable for number of known media streams instead of hardcoding in definition of sip_pvt - Rename "text" to "codecs" - beacuse it's what it is - Add documentation for future developers so that we make sure that we define new sdp media types for SRTP-variants git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Make sure we reset global_exclude_static at channel reloadOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Move capability into sip_cfg. While at it, make sure we reset it at channel ↵Olle Johansson
reload. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Move global_regcontext into the sip_cfg structureOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Move contact_ha to sip_cfg structureOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216826 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Doxygen updatesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216806 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Since it's possible to have more than 999 calls, I'm changing the call ↵Olle Johansson
counter roof to something higher. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07add doxygen and remove duplicate declaration of variableOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07After many years, remove VOCAL_DATA_HACK definitionOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Remove unneeded header files (tested on Linux and OS/X)Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Don't send MESSAGE with sendtext() if recepient doesn't allow MESSAGE requestsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Add some doxygenOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Fix typoOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07If there is no session timer in the INVITE, set it to default value (not ↵Olle Johansson
unset minimum = -1) Patch by oej closes issue #15621 Reported by: fnordian Tested by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-07Simplify the code in this functionOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04sip peer matching by address only with TCP/TLSDavid Vossel
This patch removes the contact header matching logic and adds logic to match all tcp/tls connections by ip only. Thanks to oej for finding the issue and suggesting solutions. Review: https://reviewboard.asterisk.org/r/354/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04Merged revisions 216430 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-04Do not treat every SIP peer as if they were configured with insecure=port.Russell Bryant
There was a problem in the function responsible for doing peer matching by IP address and port number such that during the second pass for checking for a peer configured with insecure=port, it would end up treating every peer as if it had been configured that way. These changes fix the logic in the peer IP and port comparison callback to handle insecure=port checking properly. This problem was introduced when SIP peers were converted to astobj2. Many thanks to dvossel for noticing this while working on another peer matching issue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Lets try not to use C++ keywords for variable names.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Added detection DTMF CID without polarity change alert.Doug Bailey
Added detection of DTMF tone energy levels on FXO channels in chan_dahdi monitoring loop so DTMF CID can be detected without the need of a polarity change precursor. (closes issue #9096) Reported by: fleed Patches: 9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819) Tested by: cyberplant, sum, maturs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Merge code associated with AST-2009-006David Vossel
(closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Add known internal IP address when autodomain=yesOlle Johansson
(closes issue #14573) Reported by: pj Patches: sip-internip-autodomain1.diff uploaded by mnicholson (license 96) modified by oej Tested by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-03Default the callback extension to "s". This is a regression.Tilghman Lesher
(closes issue #15764) Reported by: elguero Change-type: bugfix git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-02Merged revisions 215682 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r215682 | twilson | 2009-09-02 16:41:22 -0500 (Wed, 02 Sep 2009) | 18 lines Re-send non-100 provisional responses to prevent cancellation From section 13.3.1.1 of RFC 3261: If the UAS desires an extended period of time to answer the INVITE, it will need to ask for an "extension" in order to prevent proxies from canceling the transaction. A proxy has the option of canceling a transaction when there is a gap of 3 minutes between responses in a transaction. To prevent cancellation, the UAS MUST send a non-100 provisional response at every minute, to handle the possibility of lost provisional responses. (closes issue #11157) Reported by: rjain Tested by: twilson Review: https://reviewboard.asterisk.org/r/315/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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