asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	433c262a18	Merged revisions 189350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 \| file \| 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) \| 10 lines 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/branches/1.6.2@189353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	4d44738f4c	Merged revisions 188647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188647 \| dvossel \| 2009-04-15 17:10:04 -0500 (Wed, 15 Apr 2009) \| 18 lines 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/branches/1.6.2@189208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	7efc9a7297	Merged revisions 189204 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189204 \| dvossel \| 2009-04-17 20:28:45 -0500 (Fri, 17 Apr 2009) \| 18 lines 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/branches/1.6.2@189207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	d45d1b0595	Merged revisions 189137 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r189137 \| rmudgett \| 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009) \| 17 lines 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/branches/1.6.2@189140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	c70c0f1df0	Merged revisions 189097 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 \| mmichelson \| 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) \| 13 lines 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/branches/1.6.2@189105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	da10031a4a	Merged revisions 188947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188947 \| file \| 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) \| 22 lines 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/branches/1.6.2@188950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	a7ffc1e177	Merged revisions 188938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188938 \| file \| 2009-04-17 11:26:53 -0300 (Fri, 17 Apr 2009) \| 11 lines 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/branches/1.6.2@188941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3dfd1a2620	Merged revisions 188836 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r188836 \| tilghman \| 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009) \| 14 lines 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/branches/1.6.2@188839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	7e57a6e9bc	Merged revisions 188247 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 \| file \| 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) \| 7 lines 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. Upon further inspection this should make the behaviour of all other uses of the outbound proxy in the code. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@188259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	3cc7790a13	Merged revisions 188067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 \| file \| 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) \| 10 lines 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/branches/1.6.2@188070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	988c7fbbd3	Merged revisions 187906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 \| jpeeler \| 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) \| 12 lines 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/branches/1.6.2@187916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	29aec866aa	Merged revisions 187674 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 \| tilghman \| 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) \| 4 lines Ensure pvt is not NULL before dereferencing it. (closes issue #14784) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@187679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	f686ad62f6	Merge revision 187488 from trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@187564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	a8163e6d56	Merged revisions 185846 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185846 \| dvossel \| 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) \| 16 lines 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/branches/1.6.2@187531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b80c5c76fd	Merged revisions 187381 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 \| tilghman \| 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) \| 4 lines 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/branches/1.6.2@187391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	ae38d8586b	Merged revisions 187363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r187363 \| tilghman \| 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) \| 10 lines 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/branches/1.6.2@187366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	8251b08727	Merged revisions 186928 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186928 \| russell \| 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) \| 13 lines 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/branches/1.6.2@186929 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f0ee6eca57	Merged revisions 186899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 \| tilghman \| 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) \| 2 lines Add lastms to the require API call. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@186901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	fbac8a9a57	Merged revisions 186837 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 \| mmichelson \| 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) \| 7 lines Fix bad merge from fix for issue 13867. (closes issue #14686) Reported by: davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@186840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	df28619eca	Merged revisions 186461 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186461 \| kpfleming \| 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) \| 11 lines 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/branches/1.6.2@186469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e95d7ceebf	Merged revisions 186101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186101 \| kpfleming \| 2009-04-02 12:26:07 -0500 (Thu, 02 Apr 2009) \| 9 lines 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/branches/1.6.2@186111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f381fb5421	Merged revisions 186060 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 \| tilghman \| 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) \| 16 lines 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/branches/1.6.2@186063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ef1a48b00e	Merged revisions 185953 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185953 \| kpfleming \| 2009-04-02 08:51:44 -0500 (Thu, 02 Apr 2009) \| 11 lines 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/branches/1.6.2@185957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Brooks	caa5439962	Merged revisions 185363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185363 \| dbrooks \| 2009-03-31 11:46:57 -0500 (Tue, 31 Mar 2009) \| 44 lines 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/branches/1.6.2@185428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	9b6189b180	Merged revisions 185123 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 \| rmudgett \| 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) \| 9 lines 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/branches/1.6.2@185129 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	ac756bdcbb	Merged revisions 185122 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r185122 \| rmudgett \| 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) \| 26 lines 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/branches/1.6.2@185128 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	1e3625e133	Merged revisions 184948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184948 \| file \| 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) \| 21 lines 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/branches/1.6.2@184951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	dc3a259291	Merged revisions 184910 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184910 \| russell \| 2009-03-30 08:55:44 -0500 (Mon, 30 Mar 2009) \| 4 lines 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/branches/1.6.2@184913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	d627eb0451	Merged revisions 184838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 \| russell \| 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) \| 8 lines 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/branches/1.6.2@184839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	2616defc54	Merged revisions 184762 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 \| kpfleming \| 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) \| 12 lines 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/branches/1.6.2@184779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	fc65b48245	Merged revisions 184566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r184566 \| file \| 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) \| 16 lines 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/branches/1.6.2@184612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c2f4651a4b	Merged revisions 184339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184339 \| russell \| 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) \| 35 lines 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/branches/1.6.2@184343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	a96d378c85	Merged revisions 184280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184280 \| file \| 2009-03-25 16:22:06 -0300 (Wed, 25 Mar 2009) \| 5 lines 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/branches/1.6.2@184283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	614ea515c1	Merged revisions 184037 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r184037 \| russell \| 2009-03-24 16:40:44 -0500 (Tue, 24 Mar 2009) \| 6 lines 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/branches/1.6.2@184041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	1ef9ff6739	Merged revisions 183701 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183701 \| lmadsen \| 2009-03-23 14:06:40 -0400 (Mon, 23 Mar 2009) \| 7 lines 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/branches/1.6.2@183704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	37a03f1cbb	Merged revisions 183560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183560 \| russell \| 2009-03-20 12:00:58 -0500 (Fri, 20 Mar 2009) \| 10 lines 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/branches/1.6.2@183564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	a1a62c2319	Merged revisions 183511 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183511 \| eliel \| 2009-03-20 08:12:49 -0400 (Fri, 20 Mar 2009) \| 2 lines Remove duplicate <description> inside the xml documentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@183519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	412ae849e7	Merged revisions 183321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183321 \| tilghman \| 2009-03-19 14:17:31 -0500 (Thu, 19 Mar 2009) \| 15 lines 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/branches/1.6.2@183337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	059683e290	Merged revisions 183117 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r183117 \| mmichelson \| 2009-03-19 11:07:54 -0500 (Thu, 19 Mar 2009) \| 20 lines 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/branches/1.6.2@183122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	ce6869fd95	Merged revisions 183108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183108 \| file \| 2009-03-19 12:37:23 -0300 (Thu, 19 Mar 2009) \| 11 lines 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/branches/1.6.2@183111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	276f1c6ba9	Merged revisions 183028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r183028 \| jpeeler \| 2009-03-18 16:18:27 -0500 (Wed, 18 Mar 2009) \| 4 lines Add some code removed by mistake from commit 182722 that works around a file descriptor leak in versions of PWLib prior to 1.12.0. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@183031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	b085426cda	Merged revisions 182847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 \| russell \| 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) \| 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 \| russell \| 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) \| 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@182947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	20b2cd0eb1	Merged revisions 182722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182722 \| jpeeler \| 2009-03-17 15:47:31 -0500 (Tue, 17 Mar 2009) \| 15 lines Allow H.323 Plus library to be used in addition to the OpenH323 library Chan_h323 can now be compiled against both the previously supported versions of OpenH323 as well as the current H.323 Plus (version 1.20.2). The configure script has been modified to look in the default install location of h323 to hopefully help avoid using the environment variables OPENH323DIR and PWLIBDIR. Also, the CLI command "h323 show version" has been added which indicates which version of h323 is in use. (closes issue #11261) Reported by: vhatz Patches: asterisk-1.6.0.6-h323plus.patch uploaded by jthurman (license 614) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@182725 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	4e064fa071	Merged revisions 182408 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r182408 \| rmudgett \| 2009-03-16 20:54:53 -0500 (Mon, 16 Mar 2009) \| 8 lines OPENR2 uses an incorrect string value if the extension delimiter is not present. * Fixed OPENR2 using an incorrect string value if the extension delimiter is not present in the Dial() function. This was fixed for SS7 and PRI in trunk -r172400. * Made OPENR2 stripmsd behavior the same as the SS7, PRI, and others. * Removed trailing whitespace that appeared with OPENR2. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@182409 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	77a6840fd3	Add MFC/R2 support for chan_dahdi. This commit introduces official support for R2 signaling in chan_dahdi. The modifications to chan_dahdi, and the supporting library, LibOpenR2, were both written by Moises Silva. Many users are using this code, or a variant of it, in Asterisk 1.2, 1.4 and 1.6 in Brazil, México and Argentina. An unknown number of users (but at least 1) are using it in each of the following countries: Colombia, Nepal, Thailand, Venezuela, Perú, and probably others. To use this code, LibOpenR2 must be installed from http://www.libopenr2.org/. Information about configuration can be found in configs/chan_dahdi.conf.sample. The code committed is the most up to date version, which was being maintained in svn/asterisk/team/moy/mfcr2/. I would also like to include a Thank You to the many others that tested this code beyond those listed in this commit message. These are the names that I could find in the mantis issue. (closes issue #12509) Reported by: moy Patches: chan_zap-mfr2.patch uploaded by moy (license 222) Tested by: moy, korihor, viniciusfontes, Skarmeth, loloski, asbestoshead, titogarrido, heliocoelhojr, konsultex, ncorrare, ecarruda, rtorresduque, PTorres, ychen Review: http://reviewboard.digium.com/r/40/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	18ac659dc6	Merged revisions 182281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182281 \| dvossel \| 2009-03-16 12:47:42 -0500 (Mon, 16 Mar 2009) \| 7 lines Randomize IAX2 encryption padding The 16-32 byte random padding at the beginning of an encrypted IAX2 frame turns out to not be all that random at all. This patch calls ast_random to fill the padding buffer with random data. The padding is randomized at the beginning of every encrypted call and for every encrypted retransmit frame. Review: http://reviewboard.digium.com/r/193/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	0bf4624643	Merged revisions 182208 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182208 \| tilghman \| 2009-03-16 10:39:15 -0500 (Mon, 16 Mar 2009) \| 7 lines Fixup glare detection, to fix a memory leak of a local pvt structure. (closes issue #14656) Reported by: caspy Patches: 20090313__bug14656__2.diff.txt uploaded by tilghman (license 14) Tested by: caspy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	a6734b5a95	Fix an issue with requesting a T38 reinvite before the call is answered. The code responsible for sending the T38 reinvite did not check if an INVITE was already being handled. This caused things to get confused and the call to fail. The code now defers sending the T38 reinvite until the current INVITE is done being handled. (issue AST-191) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	87a8295303	improve a bit of suboptimal code git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	593d643d24	Merged revisions 181768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181768 \| mmichelson \| 2009-03-12 13:29:48 -0500 (Thu, 12 Mar 2009) \| 22 lines Properly send a 487 on an INVITE we have not responded to if we receive a BYE. If we receive an INVITE from an endpoint and then later receive a BYE from that same endpoint before we have sent a final response for the INVITE, then we need to respond to the INVITE with a 487. There was logic in the code prior to this commit which seemed to exist solely to handle this situation, but there was one condition in an if statement which was incorrect. The only way we would send a 487 was if the sip_pvt had no owner channel. This made no sense since we created the owner channel when we received the INVITE, meaning that the majority of the time we would never send the 487. The 487 being sent should not rely on whether we have created a channel. Its delivery should be dependent on the current state of the initial INVITE transaction. With this commit, that logic is now correctly in place. (closes issue #14149) Reported by: legranjl Patches: 14149.patch uploaded by mmichelson (license 60) Tested by: legranjl ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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