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2010-12-13Update ChangeLog and .version file. Merge fixes for CDR crash and chan_gtalk ↵lmadsen1-5/+40
fixup. git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.8.1.1@298203 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Merged revisions 294822 via svnmerge from rmudgett2-2/+10
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) | 11 lines Asterisk is getting a "No D-channels available!" warning message every 4 seconds. Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294823 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Merged revisions 294733 via svnmerge from jpeeler1-1/+15
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294734 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-09Analog lines do not transfer CONNECTED LINE or execute the interception macros.rmudgett3-321/+96
Add connected line update for sig_analog transfers and simplify the corresponding sig_pri and chan_misdn transfer code. Note that if you create a three-way call in sig_analog before transferring the call, the distinction of the caller/callee interception macros make little sense. The interception macro writer needs to be prepared for either caller/callee macro to be executed. The current implementation swaps which caller/callee interception macro is executed after a three-way call is created. Review: https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA SWP-2372 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294349 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Merged revisions 294242 via svnmerge from mnicholson1-31/+40
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov 2010) | 8 lines Go off hold when we get an empty reinvite telling us to. (closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294243 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08valgrind reported references to freed memory during a mISDN hangup collision.rmudgett1-173/+188
Bad things have been happening in chan_misdn because the chan_misdn channel private struct chan_list is not protected from reentrancy. Hangup collisions have be causing read and write accesses to freed memory. Converted chan_misdn struct chan_list to an ao2 object for its reference counting feature. ********** Removed an impediment to converting chan_list to an ao2 object. The use of the other_ch member in chan_list is shaky at best. It is set if the incoming and outgoing call legs are mISDN. The use of the other_ch member goes against the Asterisk architecture and can even cause problems. 1) It is used to disable echo cancellation. This could be bad if the call is forked and the winning call leg is not mISDN or the winning call leg is not the last mISDN channel called by the fork. The other_ch would become a dangling pointer. 2) It is used when the far end is alerting to hear the far end's inband audio instead of Asterisk's generated ringback tone. This is bad if the call is forked. You would only hear the last forked mISDN channel and it may not be ringing yet. The other_ch would become a dangling pointer if the call is later transferred. ********** JIRA SWP-2423 JIRA ABE-2614 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294125 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-05Fixed deadlock avoidance issues while locking channel when adding thebbryant1-18/+10
Max-Forwards header to a request. (closes issue #17949) (closes issue #18200) Reported by: bwg Review: https://reviewboard.asterisk.org/r/997/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@294084 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-04Fixes ringback tone on sip semi-attended transfer.dvossel1-0/+4
ABE-2168 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293924 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-04Do not output port in IPaddress for AMI sippeers.pabelanger1-1/+1
(closes issue #18248) Reported by: orn Patches: ami_sippeers.patch uploaded by pabelanger (license 224) Tested by: orn git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293887 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-03Merged revisions 293806 via svnmerge from rmudgett2-17/+17
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) | 20 lines Party A in an analog 3-way call would continue to hear ringback after party C answers. All parties are analog FXS ports. 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback) 4) C answers 5) A continues to hear ringback during the 3-way call. (All parties can hear each other.) * Fixed use of wrong variable in dahdi_bridge() that stopped ringback on the wrong subchannel. * Made several debug messages have more information. A similar issue happens if B and C are SIP channels. B continues to hear ringback. For some reason this only affects v1.8 and trunk. * Don't start ringback on the real and 3-way subchannels when creating the 3-way conference. Removing this code is benign on v1.6.2 and earlier. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293807 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-03Avoid valgrind warnings for ast_rtp_instance_get_xxx_addresstwilson1-3/+3
The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293803 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Merged revisions 293723 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500 (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) | 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293724 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-02Merged revisions 293647 via svnmerge from rmudgett2-4/+10
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500 (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) | 6 lines Make warning message have more useful information in it. Change "Unable to get index, and nullok is not asserted" to "Unable to get index for '<channel-name>' on channel <number> (<function>(), line <number>)". ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293648 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-01Analog 3-way call would not connect all parties if one was using sig_pri.rmudgett3-30/+49
Also the "dahdi show channel" would not show the correct 3-way call status. * Synchronized the inthreeway flag between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode() sign error and made take an analog sub channel enum. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293530 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-01Use ast_sockaddr_from_sin function not memcpypabelanger1-3/+6
This resolves some IAX2 registration issue report on the asterisk-users mailing list. (closes issue #18202) Reported by: pabelanger Patches: update_registry.patch.v2 uploaded by pabelanger (license 224) Tested by: pabelanger, Nic Colledge (mailing list) Review: https://reviewboard.asterisk.org/r/993 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293496 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-01Only offer codecs both sides support for directmediatwilson1-7/+17
When using directmedia, Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn't understand. (closes issue #17403) Reported by: one47 Patches: sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293493 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-30Merged revisions 293417 via svnmerge from rmudgett2-24/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500 (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29 Oct 2010) | 1 line Remove some more code that serves no purpose. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293418 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-30Merged revisions 293340 via svnmerge from rmudgett2-23/+0
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500 (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29 Oct 2010) | 1 line Remove some code that serves no purpose. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293341 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-29Modify sip_setoption to not complain about unknown options.jpeeler1-1/+0
This now behaves just like the other setoption callbacks. For the curious the offending option for the reporter was AST_OPTION_CHANNEL_WRITE which was getting passed due to a fix for chan_local in 286189. (closes issue #17985) Reported by: globalnetinc git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293305 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-26No need to define the struct if there are no users.rmudgett1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293081 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-26Allow the DAHDI driver to compile, even with a sufficiently older version of ↵rmudgett1-12/+12
libpri. Fixes our Bamboo builds. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@293046 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-25Several more defines that need to be altered for compiling against an older ↵tilghman1-4/+4
version of libpri git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292969 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-25Allow the DAHDI driver to compile, even with a sufficiently older version of ↵tilghman1-10/+10
libpri. Fixes our Bamboo builds. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292906 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-25Merged revisions 292867 via svnmerge from dvossel1-181/+225
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500 (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) | 27 lines This patch turns chan_local pvts into astobj2 objects. chan_local does some dangerous things involving deadlock avoidance. tech_pvt functions like hangup and queue_frame are provided with a locked channel upon entry. Those functions are completely safe as long as you don't attempt to give up that channel lock, but that is impossible to guarantee due to the required deadlock avoidance necessary to lock both the tech_pvt and both channels involved. In the past, we have tried to account for this by doing things like setting a "glare" flag that indicates what function should destroy the pvt. This was used in local_hangup and local_queue_frame to decided who should destroy the pvt if they collided in separate threads. I have removed the need to do this by converting all chan_local tech_pvts to astobj2. This means we can ref a pvt before deadlock avoidance and not have to worry about that pvt possibly getting destroyed under us. It also cleans up where we destroy the tech_pvt. The only unlink from the tech_pvt container occurs in local_hangup now, which is where it should occur. Since there still may be thread collisions on some functions like local_hangup after deadlock avoidance, I have added some checks to detect those collisions and exit appropriately. I think this patch is going to solve quite a bit of weirdness we have had with local channels in the past. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292868 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-22Merged revisions 292786 via svnmerge from lmadsen1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) | 13 lines Update the LDIF file for LDAP. The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems where I was doing an ldapadd to import the schema into the LDAP database, and the existing file would cause problems and ERROR messages when registering. Additional documention has been added based on feedback in the issue I'm closing. (closes issue #13861) Reported by: scramatte Patches: ldap-update.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, rgenthner ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292787 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-22Connected line is not updated when chan_dahdi/sig_pri or chan_misdn ↵rmudgett2-80/+455
transfers a call. When a call is transfered by ECT or implicitly by disconnect in sig_pri or implicitly by disconnect in chan_misdn, the connected line information is not exchanged. The connected line interception macros also need to be executed if defined. The CALLER interception macro is executed for the held call. The CALLEE interception macro is executed for the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches: abe_2589_c3bier.patch uploaded by rmudgett (license 664) abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/958/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292704 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-21Compile correctly on Linux (asterisk/localtime.h depends upon ↵tilghman1-0/+1
asterisk/autoconfig.h loading first). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292667 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-21Send CONNECT_ACKNOWLEDGE for CIS calls too.rmudgett1-0/+4
The originator of the Q.SIG call completion signaling link was not changed to the active state when the CONNECT message came in. The T309 processing would immediately kill the signaling link because it was not in the active state. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292489 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-19Add sip show peer info about crypto and remove dated commenttwilson1-0/+2
This patch adds information about the encryption setting to 'sip show peers' and removes an out-of-date comment from res_srtp.c and instead directs users to the proper documentation. (closes issue #18140) Reported by: chodorenko git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@292309 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-15Fixes peer's host port information being lost on sip reload.dvossel1-0/+3
(closes issue #18135) Reported by: lmadsen Patches: crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by: lmadsen git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291942 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-14Safer xml parsing, treat all clients the same, and better local candidate ↵dvossel1-133/+92
selection. The gtalk channel driver was doing several unsafe operations in regards to how it parsed incoming XML messages. I have cleaned that code up so it should be much safer now. We now treat all clients types the same. We have no reason to distinguish between GMAIL and GOOGLE VOICE clients anymore because they all work the same way. I also modified how the local ip is found. If no bindaddress is provided in the config file, we attempt to determine the local ip we would use to connect to google.com. If that fails, then we fall back to the ast_find_ourip() function as a last resort. Using the new method makes it much less likely that we would ever advertise a local RTP candidate as a loopback address. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291827 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-14Add the ability for ast_find_ourip to return IPv4, IPv6 or both.pabelanger4-8/+10
While testing chan_gtalk I noticed jabber was using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip() to return both IPv6 and IPv4 results. Adding a family parameter gives you the ablility to choose. Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results. Review: https://reviewboard.asterisk.org/r/973/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291758 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291655 via svnmerge from rmudgett3-94/+274
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500 (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010) | 20 lines Deadlock between dahdi_exception() and dahdi_indicate(). There is a deadlock between dahdi_exception() and dahdi_indicate() for analog ports. The call-waiting and three-way-calling feature can experience deadlock if these features are trying to do something and an event from the bridged channel happens at the same time. Deadlock avoidance code added to obtain necessary channel locks before attemting an operation with call-waiting and three-way-calling. (closes issue #16847) Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/971/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291656 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13More fixup for chan_gtalk.dvossel1-42/+64
This patch makes the xml parsing safer. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291578 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13The chan_dahdi faxdetect option only works for the first FAX call.rmudgett1-7/+11
The chan_dahdi faxdetect option only works for the first call. After that the option no longer works. The struct dahdi_pvt.callprogress member is the encoded user config setting for the callprogress and faxdetect config options. Changing this value alters the configuration for all following calls until the chan_dahdi.conf file is reloaded. * Fixed the chan_dahdi ast_channel_setoption callback to not change the users faxdetect config setting except for the current call. * Fixed the chan_dahdi ast_channel_queryoption callback to read the active DSP setting of the faxdetect option. * Made actually disable the active faxdetect DSP setting for the current call on the analog port. my_handle_dtmfup() is used for normal analog ports. dahdi_handle_dtmfup() is the legacy code and is no longer used unless in a radio mode. (closes issue #18116) Reported by: seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett (license 664) Review: https://reviewboard.asterisk.org/r/972/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revision 291504 fromrmudgett1-35/+49
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed, 13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the ast_channel. Must get the ast_channel lock before proceeding with release_chan() and release_chan_early() to hold off ast_hangup() from destroying the ast_channel. Missed this change for -r291468. JIRA ABE-2598 JIRA SWP-2317 .......... git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291507 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merge revision 291468 fromrmudgett1-72/+151
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed, 13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE --> RELEASE_COMPLETE * Add lock protection around channel list for find/add/delete operations. * Protect misdn_hangup() from release_chan() and vise versa using the release_lock. JIRA ABE-2598 JIRA SWP-2317 .......... git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291469 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-13Merged revisions 291393 via svnmerge from russell1-0/+16
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291393 | russell | 2010-10-13 10:29:21 -0500 (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines Lock pvt so pvt->owner can't disappear when queueing up a frame. This fixes a crash due to a hangup race condition. ABE-2601 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291394 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Gtalk enhancements and general code cleanup.dvossel1-134/+130
This patch includes several chan_gtalk enhancements. Two new gtalk.conf options have been added, externip and stunadd. Setting externip allows us to manually specify what the external IP address is outside of a NAT environment. Setting the stunaddr option to a valid stun server allows for that external ip to be retrieved via a STUN server automatically. This external IP is then advertised during call setup as a possible candidate. I have also attempted to clean up chan_gtalk's code so it meets our coding guidelines. During this cleanup I noticed several things that need to be done in the code and made a TODO section at the top of the file. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291192 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Move declaration closer to where now used.rmudgett1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291113 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-11Merged revisions 291110-291111 via svnmerge from rmudgett1-4/+9
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500 (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line Add missing unlock to an exception condition in reload_config(). ........ ................ r291111 | rmudgett | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit from handle_request_do() consistent. ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291112 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-08Make outbound Google Voice calls.dvossel1-3/+15
This patch allows for outbound Google Voice calls to be dialed from Asterisk using chan_gtalk. Below is an example dialstring. exten -> blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,) In this example, 'asterisk' is the jabber.conf profile configured to connect to your gmail account. In order to receive Google Voice calls make sure to enable 'allowguest=yes' in gtalk.conf. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290973 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-07Add Philippe Sultan to chan_gtalk author list.dvossel1-0/+2
Philippe has made some notable contributions to the gtalk channel driver. His name deserves to be listed amoung the authors of that file. Thanks Philippe! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290829 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-07Outbound gtalk calls now work correctly.dvossel1-1/+1
There was a problem with how the candidates were being built on an outbound call. This patch fixes that. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290828 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-06Fixes commented out code to use #if 0 instead.dvossel1-4/+6
Thanks to rmudgett for catching this! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290674 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-06Fixes gtalk outbound DTMF to work properly.dvossel1-8/+38
Outbound DTMF with gtalk needs to be done within the RTP stream. I discovered this after investigating a packet capture from the gmail client. Instead of performing jingle signaling DTMF, the gtalk servers expect all DTMF to arrive on the RTP stream using RFC2833 way of doing things. Chan_gtalk also had an issue with negotiating RTP payload type 106 for the telephony-event and then sending DTMF as payload 101. This has been resolved by always negotiating 101 as the payload type like we do everywhere else. With this patch, incoming google voice calls forwarded to Asterisk via gtalk work. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290648 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Fixes uninitialized memory problem in 'iax2 set debug peer' option.dvossel1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290506 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Fixes chan_gtalk to work with gmail clientdvossel1-96/+152
This patch was written by Philippe Sultan (phsultan). Thanks for keeping this up to date! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290479 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-05Resolves dnsmgr memory corruption in chan_iax2.dvossel1-141/+190
(closes issue #17902) Reported by: afried Patches: issue_17902.rev1.txt uploaded by russell (license 2) Tested by: afried, russell, dvossel Review: https://reviewboard.asterisk.org/r/965/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@290378 f38db490-d61c-443f-a65b-d21fe96a405b
2010-10-02Merged revisions 289798 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@289840 f38db490-d61c-443f-a65b-d21fe96a405b
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