6168 Commits (320b514b183c4bc721ae4c3024b59fbf073fc758)

Author	SHA1	Message	Date
Olle Johansson	320b514b18	Move contact_ha to sip_cfg structure ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216826 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	c20324021d	Doxygen updates ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	11574bcfcf	Since it's possible to have more than 999 calls, I'm changing the call counter roof to something higher. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	246e0852a7	add doxygen and remove duplicate declaration of variable ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	2e1d7378be	After many years, remove VOCAL_DATA_HACK definition ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	9c63a09344	Remove unneeded header files (tested on Linux and OS/X) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	5afc513ae3	Don't send MESSAGE with sendtext() if recepient doesn't allow MESSAGE requests ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	008b7a4ab8	Add some doxygen ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	e242e1b2ad	Fix typo ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	e1c711b7de	If there is no session timer in the INVITE, set it to default value (not 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	17 years ago
Olle Johansson	109cab6862	Simplify the code in this function ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216652 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	4596fdb788	sip peer matching by address only with TCP/TLS ... 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	17 years ago
Olle Johansson	98f18d56b8	Merged revisions 216430 via svnmerge from ... 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	17 years ago
Russell Bryant	ca23afaf2d	Do not treat every SIP peer as if they were configured with insecure=port. ... 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	17 years ago
Richard Mudgett	c2930434f6	Lets try not to use C++ keywords for variable names. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Doug Bailey	8430c87faa	Added detection DTMF CID without polarity change alert. ... 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	17 years ago
David Vossel	d09f9fd00a	Merge code associated with AST-2009-006 ... (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	17 years ago
Olle Johansson	6d6ce303cb	Add known internal IP address when autodomain=yes ... (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	17 years ago
Tilghman Lesher	a6ba2b64b1	Default the callback extension to "s". This is a regression. ... (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	17 years ago
Terry Wilson	f9816a6265	Merged revisions 215682 via svnmerge from ... 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	17 years ago
Richard Mudgett	595ab444af	Made chan_dahdi able to ignore incoming calls that are not in a MSN list for ISDN PTMP CPE spans. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a83cf36204	port string to int conversion using sscanf ... There are several instances where a port is parsed from a uri or some other source and converted to an int value using atoi(), if for some reason the port string is empty, then a standard port is used. This logic is used over and over, so I created a function to handle it in a safer way using sscanf(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	0a67bc6610	add Parkinglot info to sip show peer <foo> and skinny show line <foo> ... If we had this from the start, debugging the 'parking not using configured parkinglot' bug would have been easier. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Doug Bailey	eff8dd9a2f	Fix issue where DTMF CID detect was placing channels into signed linear mode ... made analog_set_linear_mode return back the mode that was being overwritten so it could be restored later. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	5537a4babe	SIP uri parsing cleanup ... Now, the scheme passed to parse_uri can either be a single scheme, or a list of schemes ',' delimited. This gets rid of the whole problem of having to create two buffers and calling parse_uri twice to check for separate schemes. Review: https://reviewboard.asterisk.org/r/343/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	d7f27e6705	like in chan_sip's sip_new skinny should copy the configured parkinglot from a line to the newly created channel. ... This makes callparking honor the configured parkinglot for skinny lines as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	b5dc4efb58	SIP support for keep-alive event ... keep-alive events are used by Sipura/Linksys for NAT keepalive. There currently don't appear to be any problems with NAT, but everytime a keep-alive event is received, Asterisk responds with a "489 Bad event". This error may indicate to a user that NAT problems exist just because this even is not supported. Now, rather than respond with an error, the packet is consumed and a "200 ok" is sent just to indicate we received the packet. (issue #15084) Patches: chan_sip.keepalive.v1.diff uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	7e7081439a	Honor configured parkinglot when parking and retrieving parked calls ... Thank oej for pointing out the fact that sip_new did not copy parkinglot from the peer into the newly created channel. (closes issue #15538) Reported by: gracedman Patches: 2009090100_sipnewparkinglot-161.diff.txt uploaded by mvanbaak (license 7) With mod by me to also fix callparking as well (this uploaded patch only fixed retrieving a parked call) Tested by: gracedman, mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	2cfddf8cb6	Add MASTER_CHANNEL() dialplan function, as well as a useful usage. ... (closes issue #13140) Reported by: cpina Patches: 20090807__issue13140.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen Change-type: feature git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	4af7d0c949	Fix register such that lines with a transport string, but without an authuser, parse correctly. ... (AST-228) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	8c56b871de	Removing whitespace that causes red dots in reviewboard ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	afe7034e19	Merged revisions 214940 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214940 | tilghman | 2009-08-31 11:16:52 -0500 (Mon, 31 Aug 2009) | 7 lines Also unlock the "other" channel, when returning, due to glare. (closes issue #15787) Reported by: tim_ringenbach Patches: chan_local.diff uploaded by tim ringenbach (license 540) Tested by: tim_ringenbach ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	f199054c88	Move discardremoteholdretrieval test so it applies only to the specific notification indicator values. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	552b1aa17d	Typo fix ("SIP/2.0 XXX" is 11 chars, not 10) ... (closes issue #15362) Reported by: klaus3000 Patches: chan_sip.c_logmessagefix_patch.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@214199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Moises Silva	3b1682bfe5	improve handling of openr2_chan_disconnect_call API failure, unlikely, but happened on openr2 library bug ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	4ae5535d8f	Update configure script for libpri COLP feature dependency requirements. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c28fb2bf19	Clarifying comments in sip_register, and removing a dead section ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	06ff8023f5	Register request line contains wrong address when user domain and register host differ ... (closes issue #15539) Reported by: Nick_Lewis Patches: chan_sip.c-registraraddr.patch uploaded by Nick (license 657) register_domain_fix_1.6.2 uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	1f81e544c0	fixes sip register parsing when user@domain is used ... (issue #15008) (issue #15672) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Moises Silva	1c14bd4bfd	increment the mfcr2 monitor count when clearing the call request ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213454 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Moises Silva	8a2302e118	fixed bug caused by calling ast_request without calling ast_call on an R2 channel, ie, CHANISAVAIL ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	3028e257bb	Better parsing for the "register" line ... Allows characters that are otherwise used as delimiters to be used within certain fields (like the secret). (closes issue #15008, closes issue #15672) Reported by: tilghman Patches: 20090818__issue15008.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	68b255eedc	If we have realtime caching enabled, 'sip reload' must purge users/peers, even if the config files haven't changed. ... (closes issue #12869) Reported by: bcnit Patches: 20090819__issue12869__2.diff.txt uploaded by tilghman (license 14) Tested by: lasko git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	39c9838d77	Add COLP support to chan_dahdi/sig_pri. ... Add Connected Line Presentation (COLP) support to chan_dahdi/libpri as an addition to issue 8824. This is the chan_dahdi/sig_pri portion. COLP support is now available for any switch for which libpri supports COLP (currently ETSI PTP, ETSI PTMP, and Q.SIG) with this patch. (closes issue #14068) Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/340/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	66f146e8cf	Merged revisions 212727 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212727 | rmudgett | 2009-08-18 11:00:56 -0500 (Tue, 18 Aug 2009) | 1 line Removed some deadwood and added some doxygen comments. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	4c47fce62e	Correct spelling of AGENTACCEPTDTMF in chan_agent. ... (closes issue #15668) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212581 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	36b38525a1	Merged revisions 212498 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r212498 | jpeeler | 2009-08-17 11:34:56 -0500 (Mon, 17 Aug 2009) | 12 lines Fix segfault when reloading chan_misdn. If more ports were specified than configured in misdn.conf a reload would crash asterisk. The problem was the unconfigured port was using data from the previously configured port. When the data for an unconfigured port was freed a crash would result from the double free. (closes issue #12113) Reported by: agupta Patches: bug12113.patch uploaded by jpeeler (license 325) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212506 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	0d2ef8ac5c	Merged revisions 212430 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 Fix uninitialized variable causing random MWI indications. (closes issue #15727) Reported by: doda Patches: dahdi_changes.patch uploaded by doda (license 853) ........ r212430 | rmudgett | 2009-08-17 10:36:28 -0500 (Mon, 17 Aug 2009) | 1 line Fix uninitialized variable. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	e60773a298	Add braces where missing and a few whitespace fixes in sig_analog ... (closes issue #15678) Reported by: alecdavis Patches: sig_analog_mainly_braces.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	132204459c	More code that somehow got left out of sig_analog ... * confirmanswer option now respected * check and set waiting for dialtone timer * unneeded needcallerid flag removed from analog_subchannel * ss_astchan does not need to be a void pointer * swap_channels callback updated to trunk * analog_hangup now resets channel to default law git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago