asterisk

Commit Graph

Author	SHA1	Message	Date
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	14 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	14 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
Sean Bright	94ade43b56	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	7d5836ca78	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	004babb20d	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354547 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	ead2b47907	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	15d28cfdad	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	65360e2ae5	Fixes deadlocks occuring in chan_agent due to r335976 Bad locking order was added to chan_agent to prevent segfaults from having no locking in a patch by irroot. This patch addresses the bad locking order by releasing locks before getting the right locking order to stop deadlocks from occuring when doing multiple interactions with agents. (closes issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review: https://reviewboard.asterisk.org/r/1708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	ea4fa3227f	Ensure entering T.38 passthrough does not cause an infinite loop After R340970 Asterisk was still polling the RTCP file descriptor after RTCP is shut down and removed. If the descriptor happened to have data ready when the removal occured then Asterisk would go into an infinite loop trying to read data that it can never actually access. This change disables the audio RTCP file descriptor for the duration of the T.38 transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan Vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c5fc58c3a0	Restore the 'w' modifier support for ISDN spans. Dial(DAHDI/g0/1234w888) This feature also causes the sending complete ie to be sent for switch types that do not automatically send the ie. (EuroISDN/ETSI) The main difference between dialing Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the sending of the sending complete ie. (closes issue ASTERISK-19176) Reported by: rmudgett Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ad24624751	Fix sip show peers port output, align columns, and fix ami port output. A previous patch I committed from ASTERISK-16930 unexpectedly changed some output for the AMI action "sippeers" which this patch changes back. Also, this aligns the output for the cli command "sip show peers" and fixes another issue that patch introduced by using ast_sockaddr_stringify calls multiple times without immediately using the pointer. I also went ahead and did a little janitorial work to clean up whitespace in _sip_show_peers. (issue ASTERISK-16930) (closes issue ASTERISK-19281) Reported by: Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by Walter Doekes (license 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	3c1a9894e8	Use ast_sockaddr_stringify_fmt wrappers for various functions in chan_sip There are a number of cleaner looking wrappers for ast_sockaddr_stringify_fmt available which are slightly more readable than using a direct call to ast_sockaddr_stringify_fmt. This patch switches a number of those calls in chan_sip to use those wrappers and is generally harmless. (Closes issue ASTERISK-16930) Reported by: Michael L. Young Patches: chan_sip-broken-registration-1.8.diff uploaded by Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	d699845a55	Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it anytime an address resolves to something different. There are a couple of issues with this. First, the ast_sockaddr is usually the address of an ast_sockaddr inside a refcounted struct and we never bump the refcount of those structs when using dnsmgr. This makes it possible that a refresh could happen after the destructor for that object is called (despite ast_dnsmgr_release being called in that destructor). Second, the module using dnsmgr cannot be aware of an address changing without polling for it in the code. If an action needs to be taken on address update (like re-linking a SIP peer in the peers_by_ip table), then polling for this change negates many of the benefits of having dnsmgr in the first place. This patch adds a function to the dnsmgr API that calls an update callback instead of blindly updating the address itself. It also moves calls to ast_dnsmgr_release outside of the destructor functions and into cleanup functions that are called when we no longer need the objects and increments the refcount of the objects using dnsmgr since those objects are stored on the ast_dnsmgr_entry struct. A helper function for returning the proper default SIP port (non-tls vs tls) is also added and used. This patch also incorporates changes from a patch posted by Timo Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	a61f99f985	prevent debug messsges displaying -ve Cseq numbers. Missed in R353320 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	8fc0050b54	RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers. * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. Summary of CSeq numbers. An initial CSeq number must be less than 2^31 A CSeq number can increase in value up to 2^32-1 An incrementing CSeq number must not wrap around to 0. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1699/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	2281ba7cef	Clarify log WARNING message when port-zero SDP 'm' lines received. Previously, if an m-line in an SDP offer or answer had a port number of zero, that line was skipped, and resulted in an 'Unsupported SDP media type...' warning message. This was misleading, as the media type was not unsupported, but was ignored because the m-line indicated that the media stream had been rejected (in an answer) or was not going to be used (in an offer). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a55030f4fa	Audit of ao2_iterator_init() usage for v1.8. Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as a result. Review: https://reviewboard.asterisk.org/r/1697/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352955 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	4d2f8a9cfd	rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer. If a BLF subscription exists for long enough, using %d may print negative version numbers. Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative. Tested with Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/1694/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352862 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	301cc6b1c0	Copy amaflags to sip_pvt from peer during create_addr_from_peer For whatever reason, we don't have a single function for copying data like this from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the sip_pvt, but it would probably be worth discussing this function along with the others that essentially just copy some amount of data from a peer to a private. (Closes issue ASTERISK-19029) Reported by: Matt Lehner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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