7263 Commits (98658b0670a4374be271aed0c7b42657cfced657)

Author	SHA1	Message	Date
Richard Mudgett	94fb536135	Make number not available presentation also set screening to network provided. ... Q.951 indicates that when the presentation indicator is "Number not available due to interworking" for a number then the screening indicator field should be "Network provided". * Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE when the presentation is "Number not available due to interworking". This fix makes Asterisk consistent and it also makes it consistent with earlier branches as far as this presentation value is concerned. * Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions handle the "Number not available due to interworking" case better in sig_pri.c. This change is possible because the minimum required libpri version (v1.4.11) has the necessary defines in libpri.h. ........ Merged revisions 360309 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	6b2201e80a	Add missing initialization of update_redirecting in chan_sip.c ... ........ Merged revisions 360262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	de2f65ce68	Ensure Asterisk sends a BYE when pending on the final response to a re-INVITE ... When Asterisk detects a hangup and cannot send a BYE due to a pending INVITE, it sets the pendingbye flag and waits for the final response to that INVITE. When the response is received, it transmits the BYE. If, however, that INVITE request is a pending re-INVITE, it needs to first send a CANCEL request to terminate the pending re-INVITE. In that circumstance, Asterisk was, in some scenarios, clearing the pendingbye flag after processing the CANCEL request and not checking for a pending BYE when receiving the final 487 response to the INVITE. This patch ensures that if the pendingbye flag is set, it is honored regardless of the nature of the INVITE request currently in flight. (closes issue ASTERISK-19365) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283) Review: https://reviewboard.asterisk.org/r/1807 ........ Merged revisions 360086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@360088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	b60963eab9	chan_iax2: Emit Port alongside Post in PeerStatus AMI Event. ... The PeerStatus event for IAX2 channels currently includes a header named Post which should have been Port. So include Port along with Post when emitting the event. We'll remove Post in trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	078fd3f925	Missed lastinvite CSeq int to uint32_t change ... from Review: https://reviewboard.asterisk.org/r/1699/ ........ Merged revisions 359809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	22b75fa244	chan_iax2: Fix use of uninitialized sockaddr_in in try_transfer(). ... Initialize a struct sockaddr_in in try_transfer() so that the code isn't (potentially) trying to read from it while uninitialized. ........ Merged revisions 359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	d0cb19147c	chan_gtalk: Fix potential use of uninitialized variable. ... Avoid potential use of idroster in gtalk_alloc() before it has been initialized. ........ Merged revisions 359508 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	43100ddd82	Fix deadlock potential with some ast_indicate/ast_indicate_data calls. ... Calling ast_indicate()/ast_indicate_data() with the channel lock held can result in a deadlock with a local channel because of how local channels need to avoid deadlock. ........ Merged revisions 359451 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	40cc905d86	Fix invalid reads/writes due to incorrect sizeof(). ... These few places in the code used sizeof() on h_addr in struct hostent. This is sizeof(char *). The correct way to get the size of this address is to use h_length. This error would result in reads/writes of 8 bytes instead of 4 on 64-bit machines. ........ Merged revisions 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	257238412a	Remove chan_usbradio and app_rpt. ... These modules are being maintained outside of the tree and have been for a long time now, so it doesn't make sense to keep them here. Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged revisions 359050 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@359051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	7db4fb902e	Make hints for invalid SIP devices return Unavail, not idle ... This patch drastically simplifies the device state aggegation code. The old method was not only overly complex, but also made it impossible to return AST_DEVICE_INVALID from the aggregation code. The unit test update is as a result of fixing that bug. The SIP change stems from a bug introduced by removing a DNS lookup for hostname-based SIP channels. (closes issue ASTERISK-16702) Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged revisions 358943 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ce0093e643	Make transfer not ignore port information with SIP. ... Attempting to transfer with SIP to an address like 1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from the host string and ignored. This simply keeps chan_sip from cutting off the port number during these kinds of transfers. (closes issue ASTERISK-19321) Reported by: Federico Alves Review: https://reviewboard.asterisk.org/r/1790/diff/#index_header ........ Merged revisions 358643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	830d0d5be2	Change directly setting _softhangup in sig_ss7.c to use ast_softhangup_nolock(). ... Update to: (issue ASTERISK-19372) ........ Merged revisions 358530 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3637eafa35	Fix ring cadance setup for outgoing calls on FXS ports. ... * Fix referencing the wrong variable in chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for compiling with -Wshadow and finding this bug. ........ Merged revisions 358377 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3ca4addbad	Drop SS7 call if not connected yet when INCOMPLETE/BUSY/CONGESTION. ... SS7 is a trunk protocol and should clear a failed call as soon as possible. * Made SS7 hangup a call immediately if it has not connected yet for INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate inband tone. (closes issue ASTERISK-19372) Reported by: Igor Nikolaev ........ Merged revisions 358278 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	00e7f93582	Setup DSP when SS7 call is connected or early media is available. ... Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel that requires out-of-band DTMF will not work. * Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c. The new call converts conditionaled out unconverted code and shows that the code really did something useful. * Improved some chan_dahdi DTMF debug messages to help track DTMF handling. (closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........ Merged revisions 358260 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	732eaebec7	Defer sending the connected line reinvite if a reinvite is already in progress. ... (issue ASTERISK-19355) Reported by: tomaso (closes issue AST-825) ........ Merged revisions 358162 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	e7c8651f06	Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors ... Asterisk was not setting pendinginvite in the upper half of handle_request_invite such that the 4xx was retransmitted repeatedly even though an ack was received for every retransmission. (closes issue ASTERISK-19303) Reported by: Jon Tsiros Patches: fix-19303.patch uploaded by Jeremiah Gowdy (license 6358) ........ Merged revisions 358115 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	ae3cb097ad	Fix unused-but-set-variable warnings ... All of these were pretty obviously unused. Some were unused because the code that used them was #if 0'd. In those cases, I just commented out the unused-but-set variables. ........ Merged revisions 358029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	6e64180d29	Correct some set-but-unused variable warnings in the mISDN library. ... (from kpfleming's commit to trunk r356292) ........ Merged revisions 358011 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358017 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	c2456f442e	Make chan_usbradio compile under dev mode ... x=++x and x=x=1? Really? ........ Merged revisions 357986 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f1bdfc6180	Remove ISDN hold restriction for non-bridged calls. ... The check if an ISDN call is bridged before it could be placed on hold is not necessary and is overly restrictive. The check was originally done to prevent problems with call transfers in case a user tried to transfer a call connected to an application to another call connected to an application. The ISDN transfer code has not required this restriction for quite some time because ECT could transfer any two active calls to each other. * Remove ISDN hold restriction for calls connected to applications. * Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and AST_CONTROL_UNHOLD instead of generating a warning message. (closes issue ASTERISK-19388) Reported by: Birger Harzenetter Tested by: rmudgett ........ Merged revisions 357894 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	ca5fe398a2	The default value for mohinterpret is the empty string, so when resetting to ... default values don't explicitly set the value to "default." ........ Merged revisions 357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	435ebee127	Use more reasonable cause code when rejecting incoming call waiting calls. ... (closes issue ASTERISK-19397) Reported by: Birger Harzenetter Patches: nochannel-cause.patch (license #5870) patch uploaded by Birger Harzenetter ........ Merged revisions 357407 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	6a04c78329	Add a security event for the case where fake authentication challenge is sent. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	7deeed0304	Changes transport option in sip.conf so that using multiple instances doesn't stack. ... Prior to this patch, Using "transport=" multiple times would cause them to add to one another like allow/deny. This patch changes that behavior to simply use the transport option specified last. Also, if no transport option is applied now, the default will automatically be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674) Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header ........ Merged revisions 357266 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@357271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d372c43488	Fix worker thread resource leak in SIP TCP/TLS. ... The SIP TCP/TLS worker threads were created joinable but noone could join them if they died on their own. * Fix the SIP TCP/TLS worker threads to not be created joinable. * _sip_tcp_helper_thread() only needs one parameter since the pvt parameter is only passed in as NULL and never used. (closes issue ASTERISK-19203) Reported by: Steve Davies Review: https://reviewboard.asterisk.org/r/1714/ ........ Merged revisions 356677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	e796b45951	Allow SRTP policies to be reloaded ... Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) ........ Merged revisions 356604 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3b57020680	Fix blind transfer parking issues if the dialed extension is not recognized as a parking extension. ... Custom parking extensions may not be coded such that the first and only extension priority is the Park application. These custom parking extensions will not be recognized as parking extensions. When a call is blind transferred to an extension that is not recognized as a parking extension, the normal blind transfer code causes the transferred channel to start executing dialplan. Calls that get parked in this manner do not know the original channel name that parked the call so the original parker could never be called back if the parked call is not retrieved before the timeout time. The parking space is also announced to the call being parked as a side effect of not knowing the original parking channel. * Fix handling of BLINDTRANSFER channel variable for call parking. * Fixed SIP blind transfer using the wrong dialplan context variable to check for the parking extension. (closes issue ASTERISK-19322) Reported by: aragon Tested by: rmudgett, jparker Review: https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........ Merged revisions 356521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	4501bb048f	Fix ACK routing for non-2xx responses. ... When we send an ACK for a 2xx response to an INVITE, we are supposed to use the learned route set. However, when we receive a non-2xx final response to an INVITE, we are supposed to send the ACK to the same place we initially sent the INVITE. We had been doing this up until the changes went in that would build a route set from provisional responses. That introduced a regression where we would use the learned route set under all circumstances. With this change, we now will set the destination of our ACK based on the invitestate. If it is INV_COMPLETED then that means that we have received a non-2xx final response (INV_TERMINATED indicates a 2xx response was received). If it is INV_CANCELLED, then that means the call is being canceled, which means that we should be ACKing a 487 response. The other change introduced here is setting the invitestate to INV_CONFIRMED when we send an ACK *after* the reqprep instead of before. This way, we can tell in reqprep more easily what the invitestate is prior to sending the ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer patches: ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049) (with some slight modifications prior to commit) ........ Merged revisions 356475 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	bc3cb5ddbc	Merged revisions 356214 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) | 27 lines Fix potential buffer overrun and memory leak when executing "sip show peers" The "sip show peers" command uses a fix sized array to sort the current peers in the peers ao2_container. The size of the array is based on the current number of peers in the container. However, once the size of the array is determined, the number of peers in the container can change, as the peers container is not locked. This could cause a buffer overrun when populating the array, if peers were added to the container after the array was created. Additionally, a memory leak of the allocated array would occur if a user caused the _show_peers method to return CLI_SHOWUSAGE. We now create a snapshot of the current peers using an ao2_callback with the OBJ_MULTIPLE flag. This size of the array is set to the number of peers that the iterator will iterate over; hence, if peers are added or removed from the peers container it will not affect the execution of the "sip show peers" command. Review: https://reviewboard.asterisk.org/r/1738/ (closes issue ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	34179f7093	Make 'iax2 show callnumber usage' output make sense when an IP is passed in. ... ........ Merged revisions 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@356108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	c6cde9e2c0	Remove spurious warning when 'qualifyfreqnotok' is set successfully. ... (closes issue ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John Covert (license 5512) ........ Merged revisions 355997 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	76ddf391c6	This was a LOG_NOTICE, so roll it back. ... ........ Merged revisions 355952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	fd99b4c811	Change some debug messages from LOG_DEBUG to ast_debug. ... ........ Merged revisions 355949 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	a61bda1a90	Add some boilerplate documentation for IAXVAR and IAXPEER. ... ........ Merged revisions 355904 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	4b13d1d36a	Set the length of the ast_sockaddr, so that we can set it's port later. ... Without this, the call to ast_sockaddr_set_port a few lines later is a noop. ........ Merged revisions 355901 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	b9e0c5ecf4	push 'outgoing' flag from sig_XXX up to chan_dahdi ... 'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. Now provides a callback for all the low level sig_XXX modules. (issue ASTERISK-19316) alecdavis (license 585) Reported by: Jeremy Pepper Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1747/ ........ Merged revisions 355850 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	7070d7acef	Don't allow trunkfreq to be greater than 1000ms. ... ........ Merged revisions 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	f41e7d4a44	Pass the correct value to ast_timer_set_rate() for IAX2 trunking. ... IAX2 uses the trunkfreq variable to determine how often to send trunk packets, but this value is in milliseconds while ast_timer_set_rate() expects the rate argument to be ticks per second. So we divide 1000 by trunkfreq and pass that in instead. With a default of 20ms, this change makes IAX2 send trunk packets every 20ms instead of every 50ms. Tracked down by myself and Bob Wienholt. ........ Merged revisions 355746 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	858b62ee20	Fix regressions with regards to route-set creation on early dialogs. ... This fixes two main issues: 1. Asterisk would send a CANCEL to the route created by the provisional response instead of using the same destination it did in the initial INVITE. 2. If a new route set arrives in a 200 OK than was in the 1XX response (perfectly possible if our outbound INVITE gets forked), then the route set in the 200 OK needs to overwrite the route set in the 1XX response. (closes issue ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt (license 6034) Review: https://reviewboard.asterisk.org/r/1749 ........ Merged revisions 355732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	63e4aa0740	When IAX2 debugging is enabled, make sure to log 'apathetic' messages too. ... ........ Merged revisions 355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	c1e1d5e48d	Only use maxtrunkcall and maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified. ... These variables are only accessed from the IAX_OLD_FIND path, so there is no reason to keep them updated otherwise. ........ Merged revisions 355458 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	2f84d485ee	Use TRUNK_CALL_START as originally intended. ... Back in r646, TRUNK_CALL_START was added and defined as 0x4000. That same value was also hard-coded in one part of the IAX2 code instead of using the #define. TRUNK_CALL_START has changed over the years (for dealing with LOW_MEMORY), but the hard-coded usage was never updated to match. This patch fixes that. ........ Merged revisions 355448 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	b36e1c34c6	Properly invert the return of a strncmp call. ... This was causing identification that should have been made private to be public. (closes issue AST-814) reported by Patrick Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson (license 5430) ........ Merged revisions 355268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355271 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	34d07c8706	Clear the high order bit from the destination call number before sending. ... send_apathetic_reply takes the incoming frame's source call number as the destination call number for the outgoing frame. If the incoming frame was a full frame, then the high order bit of the source call number is set and will be interpreted as a retransmit when sent back out as the destination call number. ........ Merged revisions 355182 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@355183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	cf859354d9	Fix parsing of SIP headers where compact and non-compact headers are mixed ... Change parsing of SIP headers so that compactness of the header no longer influences which header will be chosen. Previously, a non-compact header would be chosen instead of a preceeding compact-form header. (closes issue ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/ ........ Merged revisions 354702 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@354703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	06097e11b8	Clean-up of minor formatting issues in r354542/3/4 ... rmudgett pointed out some formatting issues in the check-in for ASTERISK-19290. This cleans those up. Review: https://reviewboards.asterisk.org/r/1722/ ........ Merged revisions 354547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@354548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	4f9a0e44ff	Fix SIP INFO DTMF handling for non-numeric codes ... In ASTERISK-18924, SIP INFO DTMF handlingw as changed to account for both lowercase alphatbetic DTMF events, as well as uppercase alphabetic DTMF events. When this occurred, the comparison of the character buffer containing the event code was changed such that the buffer was first compared again '0' and '9' to determine if it was numeric. Unfortunately, since the first character in the buffer will typically be '1' in the case of non-numeric event codes (10-16), this caused those codes to be converted to a DTMF event of '1'. This patch fixes that, and cleans up handling of both application/dtmf-relay and application/dtmf content types. Review: https://reviewboard.asterisk.org/r/1722/ (closes issue ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........ Merged revisions 354542 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@354543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	6fd4756e8a	Fix multiple SIP realtime issues ... 1. Set lastms to 0 when clearing instead of "" 2. Don't set ipaddr or port to the string "(null)" when they are empty 3. Add missing required fields, set default for lastms to 0, and modify the length of the ipaddr field to 45 in the Postgresql realtime.sql file. (closes issue ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/ ........ Merged revisions 354348 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@354349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago