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2010-07-16Formatting changesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Formatting fixesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277065 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Clarify syntax changesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Adding a few more to the list of CREDITSOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Formatting changes (guideline corrections)Olle Johansson
Found a unused bag of curly brackets under my table. I always wondered where they had gone. They where indeed needed in chan_sip.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Adding a few more creditsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Add ability to configure the Max-Forwards header in the dialplan, as well as inOlle Johansson
sip.conf configuration for the channel and for devices. The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary, like SIP proxys and SBCs, decrement this counter and detects when it reaches zero, at which point the SIP request is nicely killed in a SIP-friendly way. Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel for the review and good advice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Add a dialplan function to check if a queue exists: QUEUE_EXISTSOlle Johansson
Review: https://reviewboard.asterisk.org/r/777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16And yet one moreTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16"Item may be used uninitialized in this function."Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Fix reversed logic of if statement.Mark Michelson
Found based on message from Philip Prindeville on the Asterisk Developers mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Detect the --dynamic-list flag a bit betterTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Fix build on FreeBSDTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Fix trunk build for Mac OS X 10.6Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-16Allow ipaddress to contain the maximum IPv6 address.Tilghman Lesher
Also, update meetme to the full list of supported fields. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-15Quote AC_SUBST within m4_ifval, so it does not get prematurely expanded.Tilghman Lesher
(closes issue #17654) Reported by: pprindeville Patches: issue17654.diff uploaded by qwell (license 4) Tested by: qwell, pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-15Correct not setting the bindport before attempting to open the socket.Jeff Peeler
Related to changes from 276571, I was accidentally testing with a port set in my configuration causing me to miss this. Also moved the TCP handling as well to occur before build_peer is called. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-15Define LLONG_MAX on systems that do not have it.Tilghman Lesher
(closes issue #17644) Reported by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-15Fix linking asterisk on CentOS 5, which is using gcc 4.1.1. Gcc 4.1.2 has ↵Tilghman Lesher
the real fix. Review: https://reviewboard.asterisk.org/r/790/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-15Merged revisions 276652 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) | 2 lines In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-15Add lua5.1 to the handy dandy list of packages.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Fix MWI notification transmission problems over SIP.Jeff Peeler
MWI updates were not being sent if no messages were found in the event cache. This was corrected since a phone may need to clear its MWI status configured previously from another mailbox. Upon module or sip reload, MWI updates could not be sent due to the sipsock socket not being set early enough in reload_config. The code handling the descriptor assignment and such has simply been moved before the call to build_peer. Issuing a sip reload cleared the IP address of the peer, but skipped checking the database for registration information. The database is now checked both for sip reload and actually reloading the module. If a transmission occurs before the do_monitor thread has started, do not attempt to send a signal to it. (closes issue #17398) Reported by: ip-rob git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Fix errors where incorrect address information was printed.Mark Michelson
ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions) uses thread-local storage for storing the string that it creates. In cases where ast_sockaddr_stringify_fmt was being called twice within the same statement, the result of one call would be overwritten by the result of the other call. This usually was happening in printf-like statements and was resulting in the same stringified addressed being printed twice instead of two separate addresses. I have fixed this by using ast_strdupa on the result of stringify functions if they are used twice within the same statement. As far as I could tell, there were no instances where a pointer to the result of such a call were saved anywhere, so this is the only situation I could see where this error could occur. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Make compile again.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Oops, merge reverted this fix.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Remove the old stub files, preferring the optional_api method.Tilghman Lesher
(closes issue #17475) Reported by: tilghman Review: https://reviewboard.asterisk.org/r/695/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Don't try to call an embedded module's backup_globals() function untilKevin P. Fleming
after confirming it exists. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14handle special case were "200 Ok" to pending INVITE never receives ACKDavid Vossel
Unlike most responses, the 200 Ok to a pending INVITE Request is acknowledged by an ACK Request. If the ACK Request for this Response is not received the previous behavior was to immediately destroy the dialog and hangup the channel. Now in an effort to be more RFC compliant, instead of immediately destroying the dialog during this special case, termination is done with a BYE Request as the dialog is technically confirmed when the 200 Ok is sent even if the ACK is never received. The behavior of immediately hanging up the channel remains. This only affects how dialog termination proceeds for this one special case. RFC 3261 section 13.3.1.4 "If the server retransmits the 2xx response for 64*T1 seconds without receiving an ACK, the dialog is confirmed, but the session SHOULD be terminated. This is accomplished with a BYE, as described in Section 15." git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Expand the caller ANI field to an ast_party_idRichard Mudgett
Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14collapse debug code in retrans_pkt into separate linesDavid Vossel
I've been working in this function a bunch lately, and these huge debug strings are getting annoying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Make compile again.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Do not skip sending MWI for a peer if an address is defined. Really just a ↵Jeff Peeler
merge mistake from IPv6 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Fix documentation for pgsql cel and cdr, and slightly improve pgsql_cel.Tim Ringenbach
Change the documented pgsql schema to use "timestamp" instead of "time", as the latter is only a time without a date. Added some missing columns for cel's pgsql schema, and corrected spelling on some others. Updated cel's uniqueid size to be the same as the cdr. Added id column to cel's pgsql schema and updated code to allow unknown columns to get their default value instead of forcing 0 or empty string. Added microseconds to the timestamp cel logs to pgsql. Review: https://reviewboard.asterisk.org/r/734 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14ast_callerid restructuringRichard Mudgett
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Merged revisions 276267 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010) | 1 line Update documentation for voicemail.conf externpass option. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13chan_sip: RFC compliant retransmission timeoutDavid Vossel
Retransmission of packets should not be based on how many packets were sent, but instead on a timeout period. Depending on whether or not the packet is for a INVITE or NON-INVITE transaction, the number of packets sent during the retransmission timeout period will be different, so timing out based on the number of packets sent is not accurate. This patch fixes this by removing the retransmit limit and only stopping retransmission after a timeout period is reached. By default this timeout period is 64*(Timer T1) for both INVITE and non-INVITE transactions. For more information on sip timer values refer to RFC3261 Appendix A. Review: https://reviewboard.asterisk.org/r/749/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13Revert early destruction of RTP sessionsTerry Wilson
Some code improperly assumes that the sessions are still there, so revert the change until I can find all of them and fix them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13Recorded merge of revisions 276126 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276126 | russell | 2010-07-13 14:14:54 -0500 (Tue, 13 Jul 2010) | 2 lines Only reset a CDR that exists. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13Merged revisions 276123 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276123 | russell | 2010-07-13 14:06:53 -0500 (Tue, 13 Jul 2010) | 2 lines Use chan->cdr instead of chan_cdr (just like peer->cdr instead of peer_cdr in the last commit). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13Oops, XML documentation fix.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13It really cannot fail in the places below, but the stupid compiler doesn't ↵Tilghman Lesher
know that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13Weird compiler error on Bamboo.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13FILE() now supports line-mode and writing (altering) files.Tilghman Lesher
(closes issue #16461) Reported by: skyman Patches: 20100622__issue16461.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/737/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13Merged revisions 275773 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines Make user removals and traversals thread safe in meetme. Race conditions present in meetme involving the user list where a lack of locking has the potential for a user to be removed during a traversal or as in the case of the reporter after checking if the list is empty could cause a crash. Fixing this was done by convering the userlist to an ao2 container. (closes issue #17390) Reported by: Vince Review: https://reviewboard.asterisk.org/r/746/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13Destroy RTP fds when we schedule final dialog destructionTerry Wilson
Since we are only keeping the dialog around for retransmissions at this point and there is no possibility that we are still handling RTP, go ahead and destroy the RTP sessions. Keeping them alive for 32 past when they are used is unnecessary and can lead to problems with having too many open file descriptors, etc. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13Merged revisions 275994 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275994 | russell | 2010-07-13 11:51:18 -0500 (Tue, 13 Jul 2010) | 14 lines Access peer->cdr directly instead of through a saved off reference. At this point in the code, it is possible that peer_cdr may be invalid. Specifically, in the blind transfer code, CDRs are swapped between channels. So, peer_cdr is no longer == peer->cdr. The scenario that exposed a crash in this code was a blind transfer that hit the system call limit, causing the transferee channel to get destroyed after the transfer attempt failed. Even if it succeeds and this code doesn't crash, this code was still trying to reset a CDR on a channel that was now owned by a different thread, which is a BadThing(tm). (ABE-2417) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13Merged revisions 275909 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275909 | tilghman | 2010-07-13 09:47:30 -0500 (Tue, 13 Jul 2010) | 2 lines Move SQL scripts into their own database-specific directories. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-13Add example script for use with the externpasscheck voicemail.conf option.Russell Bryant
(closes issue #17628) Reported by: lmadsen Tested by: russell, lmadsen Review: https://reviewboard.asterisk.org/r/774/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-12Don't try to ref authpeer when it isn't setTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-12Add which ITU spec specifies the numbering plan.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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