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2009-03-11Merged revisions 181295 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11Fix malloc debug macros to work properly with h323.Jeff Peeler
The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11Add missing comment that quotes RFC 3891Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11Merged revisions 181029,181031 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines Remove unused variables. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-04Resolve object matching issues related to the removal of the sip_user object.Russell Bryant
Previously, chan_sip had both sip_peer and sip_user objects in memory. A patch went in to remove sip_user to simplify the code, since everything could be done with just sip_peer. This patch resolves some regressions found that were introduced by those changes. This code comes from svn/asterisk/team/group/sip-object-matching/. Here is a list of the changes that have been made: 1) When doing a match by name with the find_peer() function, make it much easier to specify which objects should be matched by having a parameter that specifies exactly which object types should be considered. Also, update find_by_name() to handle this parameter. Finally, update all code to use the new option values. 2) When looking up an object for an outbound request by name, consider peers only. (create_addr()) 3) Only match peers on an incoming registration request. 4) When doing authentication (except for SUBSCRIBE), look up users by name, instead of all objects by name. 5) When doing authentication (except for SUBSCRIBE), after looking for a user by name, look for a peer by IP address, instead of all objects by IP address. 6) When handling the SIP qualify CLI command or manager action, look for a peer by name, instead of any object by name. 7) When handling the SIP unregister CLI command, look for a peer by name, instead of any object by name. 9) In sip_do_debug_peer(), search for a peer by name, instead of any object by name. 9) When handling the SIPPEER() dialplan function, search for a peer by name, instead of any object by name. 10) In the following session timer related functions, st_get_se(), st_get_refresher(), and st_get_mode(), when looking for an object for a given sip_pvt using pvt->peername, look for a peer by name, instead of any object by name. 11) Fix build_peer() to properly handle the case where separate type=peer and type=user entries were specified in sip.conf. (closes issue #14505) Reported by: lmadsen Review: http://reviewboard.digium.com/r/172/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-04Allow for "magic" pickups to work when we wish to ignore the contextMark Michelson
When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) closes issue #14567 submitted by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03Please prefix default values with DEFAULTOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-01Properly free memory and remove scheduler entries when a transmission ↵Mark Michelson
failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24Merged revisions 178205 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-21On update, test against the existence of sipregs.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-21make chan_sip.c compile on OpenBSD again.Michiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-20Set sip_request ast_str data to NULL so ast_str_copy allocates space properlyJeff Peeler
in copy_request (issue #14478) Reported by: erik_dedecker git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18Fix ordering of output for a ChannelUpdate manager event.Joshua Colp
(closes issue #14497) Reported by: vinsik Patches: chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18T38 faxdetect should jump to the 'fax' extension for incoming calls onlyDwayne M. Hubbard
The previous implementation of T38 faxdetect resulted in both sides of the call jumping to a fax extension when both sides had 't38pt_udptl=yes' and 'faxdetect=yes' in sip.conf and a 'fax' extension in the current context. This revision will jump to a 'fax' extension on incoming calls only. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17create a UDPTL structure in create_addr_from_peer() if it does not already ↵Dwayne M. Hubbard
exist for T38 This is required to create a UDPTL structure in create_addr_from_peer() to handle the scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but is defined the peer's context. I tested this patch by enabling t38pt_udptl in the [general] section on one system and only enabling t38pt_udptl in a peer's context on the system sending a fax. Without the patch, the sending system will fail to initiate T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure". When this patch is applied the sending side will successfully initiate T38 negotiation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17In this version, we can combine the queries, because we support droppingTilghman Lesher
nonexistent columns. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17Merged revisions 176426 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines After a 'sip reload', qualifies for realtime peers weren't immediately restarted, instead waiting until the next registration. We're now caching the qualify across a reload/restart and starting the qualify immediately upon loading the peer. (closes issue #14196) Reported by: pdf Patches: 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) Tested by: pdf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16Merged revisions 176029 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16Merged revisions 175921 via svnmerge from Michiel van Baak
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-15Fix a number of problems with ast_sched_report().Russell Bryant
1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-15Merged revisions 175777 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2 lines Make sure that the debug line is not printed on debug level 0 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13Add basic (passthrough, playback, record) support for ITU G.722.1 and ↵Kevin P. Fleming
G.722.1C (also known as Siren7 and Siren14) This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12Remove useless string copy, and make sscanf safe againRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12Avoid using ast_strdupa() in a loop.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10Only decrease inringing count if above zero.Joshua Colp
(issue #13238) Reported by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10Set the type for the peer structure to be a peer as the default.Joshua Colp
(closes issue #14447) Reported by: triccyx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10Make the logic for inuse and inringing manipluation match that of 1.4. The ↵Joshua Colp
old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported. (closes issue #14399) Reported by: caspy (issue #13238) Reported by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09Fix something I messed up in the merge I just didMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09Merged revisions 174282 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06Merged revisions 174082 via svnmerge from Dwayne M. Hubbard
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06Merged revisions 173967-173968 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger ........ r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a debug message I put in by accident. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06Merged revisions 173917 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02Fix a spelling mistake.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02Add a todo. I do need to really check what's going on with this ↵Olle Johansson
kill-the-user business ;-) Why do we suddenly have two flags to set peer type? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02Small formatting changeOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02Add some well-needed improvements to the wishlist in the code, so that we ↵Olle Johansson
can close some bug reports. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-02The CID lookup feature wasn't actually working properly with dialog-info+xmlSean Bright
supporting devices. The devices (snoms, specifically) need to receive a SIP URI instead of just an extension. This adds that functionality. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29Fix "cancel answered elsewhere" through app_queue with members in chan_local.Olle Johansson
Also, implement a private cause code (as suggested by Tilghman). This works with chan_sip, but doesn't propagate through chan_local. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29- Make sure we set setvar= variables on outbound calls too, not only inbound ↵Olle Johansson
calls. - Also, change a function in app.c to return a userful value instead of always returning 0. Patch by fnordian, changed by Corydon76 and myself. This does not close the bug report, as fnordian had an additional change we're still discussing. (related to issue #14059) Reported by: fnordian Patches: chan_sip_hfield.patch uploaded by fnordian (license 110) 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14) Tested by: fnordian, Corydon76, oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29Make sure register= line supports both port and expiry at the same time.Olle Johansson
(closes issue #14185) Reported by: Nick_Lewis Patches: chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-29Merged revisions 172169 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause. This patch implements a temporary storage in the pvt and use that instead. The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header) Thanks to Klaus Darillion for testing! (closes issue #14294) related to issue #13385 Reported by: klaus3000 and adomjan Patches: bug14294b.diff uploaded by oej (license 306) Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487) Tested by: oej, klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-27Solving the same issue, but a bit different in trunk...Olle Johansson
Merged revisions 171527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines Use the same branch tag in CANCEL as in INVITE Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now. I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems. Thanks Fredrik for pointing out where the bug in the SIP messaging was. (closes issue #14346) Reported by: oej Patches: bug14346.diff uploaded by oej (license 306) Tested by: oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-26Moving generic setting to friendsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-26Continue to move variables into the sip_cfg structure to make them easier to ↵Olle Johansson
handle in the future as a group of settings for a group of devices. At some point, I want one sip_cfg per domain handled, so we can have "group" settings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-26Just moving around variable declarations so that we have all globals in the ↵Olle Johansson
same place. Default setting is set before we activate the channel or at reloads, not where we declare the variable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-26Merged revisions 171264 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes (closes issue #14284) Reported by: klaus3000 Patches: patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23Merged revisions 170504 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold. (closes issue #14295) Reported by: klaus3000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21Further fix some oddities in sip show users and sip show peers logicMark Michelson
ccesario on IRC pointed out that his sip peers were not displayed properly when he would issue the command "sip show peers." The problem was that the onlymatchonip field was used to determine if the endpoint was a "peer" or "user." The tricky part is that a "friend" is supposed to be treated as both a "user" and a "peer" but the logic would not allow "friends" to show up as "peers" since onlymatchonip was set to FALSE for friends. I have modified the sip_peer structure to more explicitly keep track of what type endpoint it is so that the various manager and CLI commands will display the expected information Reported by ccesario via IRC Tested by ccesario git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-20Convert the character pointers in a sip_request to be pointer offsetsMark Michelson
When an ast_str expands to hold more data, any pointers that were pointing to the data prior to the expansion will be pointing at invalid memory. This change makes such pointers used in chan_sip.c instead be offsets from the beginning of the string so that the same math may be applied no matter where in memory the string resides. To help ease this transition, a macro called REQ_OFFSET_TO_STR has been added to chan_sip.c so that given a sip_request and an offset, the string at that offset is returned. (closes issue #14220) Reported by: riksta Tested by: putnopvut Review http://reviewboard.digium.com/r/126/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-17Fix port :0 added to SIP INVITE URI when outboundproxy usedTerry Wilson
(closes issue #14233) Reported by: chris-mac Patches: asterisk-bug14233.diff.txt uploaded by jamesgolovich (license 176) Tested by: jamesgolovich, chris-mac, otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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