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2009-06-05Correct "dahdi show channels" output when specifying a group.Mark Michelson
Since a DAHDI channel may belong to multiple groups, we need to use a bitwise and instead of equivalence to determine whether to display the channel information. (closes issue #15248) Reported by: gentian Patches: 15248.patch uploaded by mmichelson (license 60) Tested by: gentian git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-04Merged revisions 199138 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-02fixes issue with channels not going down after transferDavid Vossel
Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop. (closes issue #15216) Reported by: oxymoron Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-02Correct documentation for the register line, specifically where the domain ↵Joshua Colp
should be specified. (closes issue #14367) Reported by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-01Tell the IAX2 parser about more control frame types.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-01Add the ability to execute connected line interception macros.Mark Michelson
When connected line updates are received or generated in the middle of an application call, it is now possible to execute a macro to manipulate the connected line data. This way, phone numbers may be manipulated to be more presentable to users, names may be changed for...whatever reason, or whatever else needs to be done may be. Review: https://reviewboard.asterisk.org/r/256 AST-165 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-01Remove extra lock from local_indicate in connected line case.Mark Michelson
Oh, and this fixes a deadlock I was seeing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-01Add missing unlock of local pvt.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-01Remove documentation for the 'exten' argument to the AGENT function.Mark Michelson
Since AgentCallbackLogin has been removed, this should not be documented any more. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-01Fix a bug where the Event and Content-Type headers were added twice to ↵Joshua Colp
outgoing SIP NOTIFY messages. (closes issue #15239) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-30When removing all packets from a dialog we also need to free the data if ↵Joshua Colp
present. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-30Remove not used code in the Agent channel.Eliel C. Sardanons
This code was there because of the AgentCallbackLogin() application. ->loginchan[] member was only used by AgentCallbackLogin(). Agent where dumped to astdb if they where logged in using AgentCallbacklogin() so they are not being dumper anymore. Review: https://reviewboard.asterisk.org/r/267/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-29New signaling module to handle analog operations in chan_dahdiJeff Peeler
This branch splits all the analog signaling logic out of chan_dahdi.c into sig_analog.c. Functionality in theory should not change at all. As noted in the code, there is still some unused code remaining that will be cleaned up in a later commit. Review: https://reviewboard.asterisk.org/r/253/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-29Fix a bug where the default setting did not perform a remote bridge when it ↵Joshua Colp
should have. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-29A few fixes to SIP with regards to connected line updates during transfers.Mark Michelson
* Set the invitestate to INV_CALLING when we send a connected line reinvite. This prevents us from potentially rapid-firing reinvites to a single peer. * Use the astdb to store a peer's allowed methods. This prevents us from sending an UPDATE during the interval between startup and the peer's first registration if the peer does not support the UPDATE method. * Handle Polycom's method of indicating allowed methods in REGISTER. Instead of using an Allow header, they place the allowed methods in a methods= parameter in the Contact header. ABE-1873 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Update references to bugs.digium.com and reviewboard.digium.com to the new URLs.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Treat 405 responses the same way we would a 501.Mark Michelson
This makes sure that we mark a method as being unallowed if we receive a 405 response so that we don't continue to try to send that same type of message. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Add missing lock to local_indicate function for connected line frames.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Fix a bug where the trunkmtu setting was not set to the default value of ↵Joshua Colp
1240 on load but was on reload. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Merged revisions 197562 via svnmerge from Eliel C. Sardanons
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines Use the address we already know when reloading a peer with nat=yes. If we already have an address for a peer, and we are reloading the sip configuration, try to use that address to contact the peer, instead of getting it from the Contact. (closes issue #15194) Reported by: ibc Patches: sip.patch uploaded by eliel (license 64) Tested by: manwe ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Eliminate several needless checks and fix a few memory leaksTilghman Lesher
(closes issue #14833) Reported by: contactmayankjain Patches: all_changes.patch uploaded by contactmayankjain (license 740) slightly modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Merged revisions 197466 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting. The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated (or it passes through unauthenticated) the proper nat flag is set. (closes issue #13823) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27Adding some generic handling of error codes sent to us in replys to requests.Olle Johansson
Previously they always set hangupcause 0, which is generally wrong. With this change, we're setting some generic hangup causes. For 5xx errors, which indicate some sort of problem with the remote server, we're now setting CONGESTION. EDVX002 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27Display an error message when chan_alsa fails to load due to a missingSean Bright
or inaccessible configuration file. Before this change, when chan_alsa failed to load due to a missing or inaccessible configuration file, no message would be displayed. With this change, when chan_alsa fails to load due to a missing or inaccessible configuration file, a message will be displayed. (closes issue #14760) Reported by: Nick_Lewis Patches: chan_alsa.c-confload.patch uploaded by Nick (license 657) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-26Remove some redundant or unnecessary connected line-related function calls.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-26Fix a bug where the sip unregister CLI command did not completely unregister ↵Joshua Colp
the peer. (closes issue #15118) Reported by: alecdavis Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-23set MFCR2_CATEGORY just when starting the pbxMoises Silva
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22SIP set outbound transport type from RegistrationDavid Vossel
In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22Don't crash if an RTP instance can't be created. This could occur when anSean Bright
invalid bindaddr was specified in gtalk.conf. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22Implement a new element in AstXML for AMI actions documentation.Eliel C. Sardanons
A new xml element was created to manage the AMI actions documentation, using AstXML. To register a manager action using XML documentation it is now possible using ast_manager_register_xml(). The CLI command 'manager show command' can be used to show the parsed documentation. Example manager xml documentation: <manager name="ami action name" language="en_US"> <synopsis> AMI action synopsis. </synopsis> <syntax> <xi:include xpointer="xpointer(...)" /> <-- for ActionID <parameter name="header1" required="true"> <para>Description</para> </parameter> ... </syntax> <description> <para>AMI action description</para> </description> <see-also> ... </see-also> </manager> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22s/it's/its/Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22resolve compiler warningRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22Fix build under dev mode and remove some casts that are no longer necessary asSean Bright
a result of the const-ify the world patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22Make chan_misdn compile.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22Merged revisions 196116 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 lines Fix a bug where using immediate with mISDN caused a cause code of 16 to get sent back instead of 1 if the 's' extension did not exist. (closes issue #12286) Reported by: lmamane ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Merged revisions 195991 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer. There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement. (closes issue #15032) Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380) Tested by: guillecabeza (closes issue #14216) Reported by: Andrey Sofronov ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-20Get rid of some duplicated code and correct a connected line error.Mark Michelson
When receiving a 200 OK response to an INVITE, it was possible to transmit two connected line updates instead of a single one. Furthermore, the second did not have the proper information present. Now the two have been combined into a single update and the correct information is presented. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-19Add basic support for handling connected line-related UPDATE requests.Mark Michelson
SIP purists may want to look the other way... When COLP/CONP support for SIP was committed, there was a condition under which Asterisk may transmit a SIP UPDATE in order to communicate the change in connected line information. The issue here is that while we could send a SIP UPDATE message, we were not prepared to receive such an UPDATE and would always responde with a 501 when we received an UPDATE. The situation was a bit rough. We really want to be able to receive UPDATEs having to do with connected line changes, but the amount of effort involved in properly supporting RFC 3311 was staggering. This commit represents a compromise. First, it was decided that it is important to only send a SIP UPDATE to an endpoint that is able to handle one. So, now we have added parsing of the Allow header into SIP. We store the allowed methods on SIP peers so that when we communicate with them, we already will know what we can and cannot send to them. We will parse the peer's allowed methods when he registers with us. If the peer is not the type to register with us, but the qualify option is enabled, then we will use the response to the OPTIONS request we send the peer to determine the peer's allowed methods. When the peer's registration expires, or when qualify deems the peer to be unreachable, we clear the allowed methods from the peer. For an actual call, we will copy the peer's allowed methods to the sip_pvt representing the call leg. If we are communicating with an endpoint which is not a peer, then we will just parse the Allow header from the first message we receive during the call and store the information in the sip_pvt. If, during communication with a peer, we receive a 501 response, then we will make sure to save the fact that we cannot use that method when communicating with that peer. Now, with all that infrastructure in place, the only actual place we use this information currently is when attempting to send a connected line change using an UPDATE request. If we cannot send the change immediately using an UPDATE, we will set the SIP_NEEDREINVITE flag so that we can send a REINVITE as soon as it is allowed. The second part of the changes here is for Asterisk to accept UPDATE requests that have connected line changes. Since we are not fully supporting RFC 3311, Asterisk will NOT place the UPDATE method in Allow headers it sends. Instead, if you are communicating with what you know to be another Asterisk box, you may set the rpid_update parameter in sip.conf so that we will send UPDATEs to that Asterisk box. When we send a connected line update, we set a custom header called "X-Asterisk-rpid-update." On the receiving end, if Asterisk receives an UPDATE that does not have the "X-Asterisk-rpid-update" header present, then Asterisk will respond with a 501 since media-changing UPDATEs are not supported. We should never get such UPDATEs, since as was stated earlier, Asterisk does not put UPDATE in its Allow header. If the custom header is present in the received UPDATE, though, then we will check the incoming request for connected line updates and queue the update on the channel where the change occurred. ABE-1840 ABE-1822 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-19Merged revisions 195448 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195448 | file | 2009-05-19 11:41:45 -0300 (Tue, 19 May 2009) | 7 lines Fix a bug where direct RTP setup would partially occur even when disabled if the calling channel was answered. (issue #13545) Reported by: davidw (issue #14244) Reported by: mbnwa ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18The facilityenable parameter does not have anything to do with pritimer ↵Richard Mudgett
parameters. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Fix a bug where specifying an empty outboundproxy would cause packets to get ↵Joshua Colp
sent to ourself. (closes issue #15106) Reported by: timeshell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Merged revisions 194873 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194873 | dvossel | 2009-05-15 17:43:13 -0500 (Fri, 15 May 2009) | 17 lines IAX2 REGAUTH loop IAX was not sending REGREJ to terminate invalid registrations. Instead it sent another REGAUTH if the authentication challenge failed. This caused a loop of REGREQ and REGAUTH frames. (Related to Security fix AST-2009-001) (closes issue #14867) Reported by: aragon Tested by: dvossel (closes issue #14717) Reported by: mobeck Patches: regauth_loop_update_patch.diff uploaded by dvossel (license 671) Tested by: dvossel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194874 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Merged revisions 194557,194685 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194557 | dvossel | 2009-05-14 17:59:43 -0500 (Thu, 14 May 2009) | 10 lines IAX2 "Ghost" Channels There is a bug tracker issue where people are reporting "Ghost" channels in their 'iax2 show channels' output. The confusion is caused by channels being listed as "(NONE)" with format "unknown". These are not channels of coarse. They are usually just pending registration or poke requests, but it is confusing output. To help make sense of this I have added two columns to 'iax2 show channels'. One shows the first message which started the transaction, and the second shows the last message sent by either side of the call. This helps diagnose why the entry exists and why it may not go away. (closes issue #14207) Reported by: clive18 Review: https://reviewboard.asterisk.org/r/246/ ........ r194685 | dvossel | 2009-05-15 10:40:37 -0500 (Fri, 15 May 2009) | 6 lines Update to previous IAX2 "Ghost" Channels patch. Fixed some comments made on reviewboard for the previous patch. (issue #14207) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14Merged revisions 194484 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines Fix a race condition where a reinvite could trigger a 482 response. The loop detection/spiral detection code in chan_sip used the owner channel's state as a criterion for determining if the incoming INVITE is a looped request. The problem with this is that the INVITE-handling code happens in a different thread than the thread that marks the owner channel as being up. As a result, if a reinvite were to come in very quickly, say from another Asterisk on the same LAN, it was possible for the reinvite to arrive before the owner channel had been set to the up state. This patch corrects the problem by using the invitestate of the sip_pvt instead, since that can be guaranteed to be set correctly by the time the reinvite arrives. Since there is a switch statement further in the INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate of the sip_pvt in case we should actually be treating the channel as if it were up already. (closes issue #12215) Reported by: jpyle Patches: 12215_confirmed.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14Add outgoing_colp misdn.conf port parameter.Richard Mudgett
Select what to do with outgoing COLP information on this port. 0 - Send out COLP information unaltered. (default) 1 - Force COLP to restricted on all outgoing COLP information. 2 - Do not send COLP information. outgoing_colp=0 Also fixed sending the EctInform message so it always has the required redirectionNumber parameter when the status is active. JIRA ABE-1853 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-12Update spiral support in trunk and 1.6.X to match what is in 1.4.Mark Michelson
In 1.4, a SIP spiral is treated the same way as a call forward. This works much better than what is currently in trunk and 1.6.X. The code in trunk and 1.6.X did not create a new call to the recipient of the spiral, instead trying to continue the same call. In addition to just being plain wrong, this also had the side effect of only being able to spiral calls to other SIP channels. With this in place, as long as call forwards are honored, SIP spirals will work properly. This means that it will work for outbound calls made by the Queue, Dial, and Page applications. For originated calls and spool calls, however, the spiral will not work properly until a generic call forward mechanism is introduced into Asterisk. (relates to issue #13630) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-12add 'const' qualifiers in various places where they should have beenKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-11Merged revisions 193613 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines Sent wrong message to clear a call we started if the other end has not responed yet. In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet), it is not allowed to clear the call with RELEASE_COMPLETE. It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-08TCP not matching valid peer.David Vossel
find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it. Review: http://reviewboard.digium.com/r/236/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193387 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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