7173 Commits (5fcb78840f128df293ba23e973a9d80da1173b4e)

Author	SHA1	Message	Date
Matthew Jordan	5c318b19c2	Fix a typo in the warning messages for an ignored media stream ... Added a '\n' to the warning messages when we ignore a media stream due to the port number being '0'. (closes issue ASTERISK-19646) Reported by: Badalian Vyacheslav git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	ed76cdda72	Replace GNU old-style field designator extensions to fix clang warnings ... (issue ASTERISK-19540) Reported by: Makoto Dei Patches: clang-gnu-designator.patch uploaded by Makoto Dei (license 5027) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	063aa93c46	Stop sending out RTCP if RTP is inactive ... This change prevents Asterisk from sending RTCP receiver reports during a remote bridge since it is no longer receiving media and should not be reporting anything. (related to ASTERISK-19366) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	58565827e6	Improve accuracy of identifying information sent in dialog-info SIP NOTIFY requests. ... This change makes use of connected party information in addition to caller ID in order to populate local and remote XML elements in the dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by: Maciej Krajewski Tested by: Maciej Krajewski Patches: local_remote_hint2.diff uploaded by Mark Michelson (license 5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	d58fe85724	Destroy configs when they are no longer used ... https://reviewboard.asterisk.org/r/1834/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	390e8c1f73	Make a debug message regarding subscription changes more accurate. ... I was getting confused during some testing why Asterisk was saying that a subscription was being added when it was clearly being removed. This fixes that confusion. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c04afdbc70	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f9246f83c9	Add missing initialization of update_redirecting in chan_sip.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	cfdc12387b	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	1ded276c97	Missed lastinvite CSeq int to uint32_t change ... from Review: https://reviewboard.asterisk.org/r/1699/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	167dd53f80	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	ba3230cc45	chan_gtalk: Fix use of uninitialized vars in config handling. ... Fix potential use of context, parkinglot, and prefs before they are initialized. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	e470c22ed6	chan_gtalk: Fix potential use of uninitialized variable. ... Avoid potential use of idroster in gtalk_alloc() before it has been initialized. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	5b61050209	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	08e1d392f4	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	0da9d71905	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	7622a34c89	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	8e2e2bf059	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fff4e1ca50	Change directly setting _softhangup in sig_ss7.c to use ast_softhangup_nolock(). ... Update to: (issue ASTERISK-19372) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	0c315bb90a	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	4c9168a4c7	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e0c235bd9b	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	ffa247ce6c	Defer sending the connected line reinvite if a reinvite is already in progress. ... (issue ASTERISK-19355) Reported by: tomaso (closes issue AST-825) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	fea6466555	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	d9961b2768	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	7495f69c95	Correct some set-but-unused variable warnings in the mISDN library. ... (from kpfleming's commit to trunk r356292) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	b686d4785f	Make chan_usbradio compile under dev mode ... x=++x and x=x=1? Really? git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	a36c4234a4	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	b3fb9153dd	The default value for mohinterpret is the empty string, so when resetting to ... default values don't explicitly set the value to "default." git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ec57a80169	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	52c50e4da7	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	534213a074	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	032962f1a2	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f49ff3ff9c	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	775b218b35	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	6453352768	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/1.8@356214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	e880b4a205	Make 'iax2 show callnumber usage' output make sense when an IP is passed in. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356107 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	cb8d4a1d50	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	a8989c5ded	This was a LOG_NOTICE, so roll it back. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	11991e8394	Change some debug messages from LOG_DEBUG to ast_debug. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	4b59946c41	Add some boilerplate documentation for IAXVAR and IAXPEER. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	3925b8fdc9	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	1b6601bc0a	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	0106636e42	Don't allow trunkfreq to be greater than 1000ms. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	338fd29f44	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	202d83c42c	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	dfe4ff5337	When IAX2 debugging is enabled, make sure to log 'apathetic' messages too. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	314dcc01bc	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355458 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	62b7e35b71	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355448 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	a5d76e1c11	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago