asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	8ba3de43ad	res_pjsip_nat: Adjust when contact should be rewritten. A previous change made the contact only get rewritten if the dialog's route set was not marked frozen. Unfortunately, while the intent of this is correct, the dialog's route set actually gets marked as frozen earlier than expected, especially for UAS dialogs. Instead, the idea is that the contact needs to not be rewritten if there is a pre-existing route set on the dialog. This is now accomplished by checking the dialog's route set list instead of checking if the route set is frozen. Doing this causes some broken tests to begin passing again. ASTERISK-25196 Reported by Mark Michelson Change-Id: I525ab251fd40a52ede327a52a2810a56deb0529e	10 years ago
Mark Michelson	56077bb1b0	Merge "res_pjsip_mwi: Set up unsolicited MWI upon registration." into certified/13.1	10 years ago
Mark Michelson	20f50131d7	res_pjsip_refer: Prevent sending duplicate headers. res_pjsip_refer will attempt to add Referred-By or Replaces headers to outbound INVITEs at times. If the INVITE gets challenged for authentication, then we will resend the INVITE. Prior to this patch, the Referred-By or Replaces header would be re-added to the outbound INVITE, resulting in duplicated headers. ASTERISK-25204 #close Reported by Mark Michelson Change-Id: I59fb5c08b4d253c0dba9ee3d3950b5025358222d	10 years ago
Matt Jordan	3ba48755c8	Merge "AMI: Add Linkedid to the standard channel snapshot AMI event headers." into certified/13.1	10 years ago
Mark Michelson	0d535df734	res_pjsip_nat: Rewrite route set when required. When performing some provider testing, the rewrite_contact option was interfering with proper construction of a route set when sending an ACK after receiving a 200 OK response to an INVITE. The initial INVITE was sent to address sip:foo. The 200 OK had a Contact header with URI sip:bar. In addition, the 200 OK had Record-Route headers for sip:baz and sip:foo, in that order. Since the Record-Route headers had the lr parameter, the result should have been: * Set R-URI of the ACK to sip:bar. * Add Route headers for sip:foo and sip:baz, in that order. However, the rewrite_contact option resulted in our rewriting the Contact header on the 200 OK to sip:foo. The result was: * R-URI remained sip:foo. * We added Route headers for sip:foo and sip:baz, in that order. The result was that sip:bar was not indicated in the ACK at all, so the far end never received our ACK. The call eventually dropped. The intention of rewrite_contact is to rewrite the most immediate destination of our SIP request to be the same address on which we received a request or response. In the case of processing a SIP response with Record-Route headers, this means that instead of rewriting the Contact header, we should instead rewrite the bottom-most Record-Route header. In the case of processing a SIP request with Record-Route headers, this means we rewrite the top-most Record-route header. Like when we rewrite the Contact header, we also ensure to update the dialog's route set if it exists. ASTERISK-25196 #close Reported by Mark Michelson Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f	10 years ago
Richard Mudgett	3332869b48	AMI: Add Linkedid to the standard channel snapshot AMI event headers. ASTERISK-25189 #close Reported by: John Hardin Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac	10 years ago
Joshua Colp	a35d6feae2	res_pjsip_mwi: Set up unsolicited MWI upon registration. The res_pjsip_mwi previously required a reload to set up the proper subscriptions to allow unsolicited MWI to work. This change makes it so the act of registering will also cause this to occur. This is particularly useful if realtime is involved as no reload needs to occur within Asterisk to cause the MWI information to get sent. ASTERISK-25180 #close Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252	10 years ago
Matt Jordan	3e3ea112c6	Merge "bridge: When performing a blonde transfer update connected line information." into certified/13.1	10 years ago
Joshua Colp	75589c4a3b	bridge: When performing a blonde transfer update connected line information. When performing a blonde transfer the code uses the old masquerade mechanism to move a channel around. As a result of this certain information, such as connected line, is moved between the channels involved. Upon completion of the move a frame is queued which is supposed to update the connected line information on the channel. This does not occur as the code considers it a redundant update since the masquerade operation updated the channel (but did not inform it of the new connected line information). The code also does not queue a connected line update to be handled by the thread handling the channel. Without this any other channel that may be loosely involved does not know it is talking to a different caller. This change does the following to resolve this: 1. The indicated connected line information is cleared upon completion of the masquerade operation when doing a blonde transfer. This prevents the connected line update from being considered redundant. 2. A connected line update frame is now queued upon the completion of the masquerade operation so any other channel loosely involved knows that there is a different caller. ASTERISK-25157 #close Reported by: Joshua Colp Change-Id: Ibb8798184a1dab3ecd35299faecc420034adbf20	10 years ago
Richard Mudgett	8142b922ab	app_directory: Fix crash when using the alias option 'a'. The voicemail.conf mailbox key/value pair is defined as: <mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]] Where all fields in the value including the field values are optional. Since the parsing code for the mailbox key/value pair is sloppy, this patch tightens the parsing for the directory information. * Renamed the 'pos' and 'bufptr' variables to 'name' and 'options' respectively in search_directory_sub(). Those names make more sense. * Made sure that search_directory_sub() is dealing with the voicemail.conf mailbox options field if it even exists when looking for the 'hidefromdir' and 'alias' options. * Fix crash if a voicemail.conf mailbox is just <mailbox>=<password>,<name> when the 'a' option is used. If there were no fields after the name then the 'options' pointer was not checked for NULL. * Fix users.conf alias processing if the 'a' option is used. The wrong variable was used. ASTERISK-25087 #close Reported by: Chet Stevens Change-Id: I86052ea77307beddddba5279824d39dc0d593374	10 years ago
Joshua Colp	2e04bc2e6d	Merge "AMI: Escape string values." into certified/13.1	10 years ago
Matt Jordan	ca2174bb23	.version: Update for certified/13.1-cert3-rc1	10 years ago
Matt Jordan	2ef2c12fae	.lastclean: Update for certified/13.1-cert3-rc1	10 years ago
Matt Jordan	5032390639	realtime: Add database scripts for certified/13.1-cert3-rc1	10 years ago
Kevin Harwell	2bf6fd263a	AMI: Escape string values. So this issue is a bit complicated. Since it is possible to pass values to AMI that contain a '\r\n' (or other similar sequences) these values need to be escaped. One way to solve this is to escape the values and then pass the escaped values to the AMI variable parameter string building function. However, this puts the onus on the pre-build function to escape all string values. This potentially requires a fair amount of changes along with a lot of string allocations/freeing for all values. Surely there is a way to push this complexity down a level into the string building function itself? This of course is possible, but ends up requiring a way to distinguish between strings that need to be escaped and those that don't. The best way to handle this is by introducing a new format specifier in the format string. For instance a %s (no escape) and %S (escape). However, that is a bit weird and unexpected. So faced with those possibilities this patch implements a limited version of the first option. Instead of attempting to escape all string values this patch only escapes those values that make sense. This approach limits the number of changes and doesn't suffer from the odd format specifier problem. ASTERISK-24934 #close Reported by: warren smith Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0	10 years ago
Mark Michelson	5f954e1e00	res_pjsip: Prevent access of NULL channels. It is possible to receive incoming requests or responses after the channel on an ast_sip_session has been destroyed and NULLed out. Handlers of these sorts of requests or responses need to be prepared for the possibility that the channel is NULL or else they could cause a crash. While several places have been amended to deal with NULL channels, there were still a couple of places that needed updating. res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to return early if there is no channel on the session. res_pjsip_session.c: When handling a 302 response, we need to stop the redirecting attempt if there is no channel on the session. ASTERISK-25148 #close reported by Mark Michelson Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9	10 years ago
Richard Mudgett	c994a3bfa0	res_pjsip_refer: Fix crash from a REFER and BYE collision. Analyzing a one-off crash on a busy system showed that processing a REFER request had a NULL session channel pointer. The only way I can think of that could cause this is if an outgoing BYE transaction overlapped the incoming REFER transaction in a collision. Asterisk sends a BYE while the phone sends a REFER to complete an attended transfer. * Made check the session channel pointer before processing an incoming REFER request in res_pjsip_refer. * Fixed similar crash potential for res_pjsip supplement incoming request processing for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE, res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail REFER messages. * Made res_pjsip_messaging respond to a message body too large with a 413 instead of ignoring it. ASTERISK-24700 #close Reported by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4417/ ........ Merged revisions 431898 from http://svn.asterisk.org/svn/asterisk/branches/13 Change-Id: I57878adc0846dd942a699ad36dcec9cba5e57994	10 years ago
Matt Jordan	a159e08d15	Merge "res_pjsip_session: Fix in-dialog authentication." into certified/13.1	10 years ago
Kevin Harwell	1e98fcac6b	res_pjsip: config option 'timers' can't be set to 'no' When setting the configuration option 'timers' equal to 'no' the bit flag was not properly negated. This patch clears all associated flags and only sets the specified one. pjsip will handle any necessary flag combinations. Also went ahead and did similar for the '100rel' option. ASTERISK-24910 #close Reported by: Ray Crumrine Review: https://reviewboard.asterisk.org/r/4582/ ........ Merged revisions 434131 from http://svn.asterisk.org/svn/asterisk/branches/13 Change-Id: Ibbc25d4592aabf7596ef473447d630961f88c217	10 years ago
Richard Mudgett	bd32327353	res_pjsip_session: Fix in-dialog authentication. When the remote peer requires authentication for in-dialog requests then re-INVITEs to the peer cause the call to be disconnected and other in-dialog requests to the peer like MESSAGE just don't go through. * Made session_inv_on_tsx_state_changed() handle in-dialog authentication for re-INVITEs and other methods. Initial INVITEs cannot be handled here because the INVITE transaction must be restarted earlier. * Pulled needed code from res/res_pjsip/pjsip_outbound_auth.c in preparation for removing the file. The generic outbound authentication code did not work as well as anticipated. * Created outbound_invite_auth() to only handle initial outbound INVITEs. Re-INVITEs cannot be handled here. The re-INVITE transaction is still in progress and the PJSIP library cannot handle the overlapping INVITE transactions. Other method types should not be handled here as this code only works on outgoing calls and we need to handle incoming and outgoing calls. ASTERISK-25131 #close Reported by: Richard Mudgett Change-Id: I12bdd7ddccc819b4ce4b091e826d1e26334601b0	10 years ago
Jonathan Rose	b81353a0ec	app_voicemail: fix moving when old messages full When completing voicemail playback of a message in the 'INBOX', the message gets moved to the 'Old' messages folder. Without this patch, if the 'Old' folder is already at its set limit, then the 'INBOX' message will simply be deleted. With this patch, the flag to delete the message will be removed if the save_to_folder function indicates that the message could not be moved due to a full folder. ASTERISK-25082 #close Reported by: Jonathan Rose Review: https://gerrit.asterisk.org/#/c/448/ Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f	10 years ago
Richard Mudgett	523fab02d8	chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision. If an ISDN call is hungup by both sides at the same time a crash could happen. * Added missing NULL checks for the owner channel after calling pri_queue_pvt_cause_data() in two places. Code after those calls need to check the owner channel pointer for NULL before use because pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the owner and the owner may get hung up. ASTERISK-21893 #close Reported by: Alexandr Gordeev Change-Id: Ica3e266ebc7a894b41d762326f08653e1904bb9a	10 years ago
Joshua Colp	dd1363a510	Merge "bridge.c: NULL app causes crash during attended transfer" into certified/13.1	10 years ago
Kevin Harwell	b764454d4d	bridge.c: NULL app causes crash during attended transfer Due to a race condition there was a chance that during an attended transfer the channel's application would return NULL. This, of course, would cause a crash when attempting to access the memory. This patch retrieves the channel's app at an earlier time in processing in hopes that the app name is available. However, if it is not then "unknown" is used instead. Since some string value is now always present the crash can no longer occur. ASTERISK-24869 #close Reported by: viniciusfontes Review: Change-Id: I5134b84c4524906d8148817719d76ffb306488ac	10 years ago
Joshua Colp	6433b697ae	res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination The res_pjsip_exten_state module currently has a race condition between processing the extension state callback from the PBX core and processing the subscription shutdown callback from res_pjsip_pubsub. There is currently no synchronization between the two. This can present a problem as while the SIP subscription will remain valid the tree it points to may not. This is in particular a problem as a task to send a NOTIFY may get queued which will try to use the tree that may no longer be valid. This change does the following to fix this problem: 1. All access to the subscription tree is done within the task that sends the NOTIFY to ensure that no other thread is modifying or destroying the tree. This task executes on the serializer for the subscriptions. 2. A reference to the subscription serializer is kept to ensure it remains valid for the lifetime of the extension state subscription. 3. The NOTIFY task has been changed so it will no longer attempt to send a NOTIFY if the subscription has already been terminated. ASTERISK-25057 #close Reported by: Matt Jordan Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643	10 years ago
Joshua Colp	5602fc3e2e	Merge "res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions." into certified/13.1	10 years ago
Joshua Colp	bf31a486cb	res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions. The first thing this patch fixes is UAS dialogs. Previously if a transport was configured on an endpoint and an inbound session was created there was no guarantee that requests sent on the dialog would use the correct transport and address information. This has now been fixed so an explicitly configured transport is taken into account. The second thing this patch fixes is res_pjsip_multihomed. The res_pjsip_multihomed module attempts to determine what transport a message should go out on and what addressing information should go into the message itself. In a scenario where multiple transports exist bound to the same IP address but a different port the code would incorrectly alter the transport and change the message to the wrong transport. This change makes the res_pjsip_multihomed module smarter so it will only change the transport and address information in the message when it is possible and makes sense. ASTERISK-24615 #close Reported by: David Justl Change-Id: I5b57362201cc8c6555834ec8707e9fbddeff7904	10 years ago
Joshua Colp	7c687c8e54	stasis: Fix dial masquerade datastore lifetime A recent change went into Asterisk which added reference counts to the channels stored in a dial masquerade datastore. Unfortunately this included a reference to the caller in a dialing operation. While all of the dialed targets have the datastore removed from them upon dialing completion this did not occur for the caller, causing it to have a reference to itself that could go never go away (as it depended on the destruction of the datastore which only happened when the channel was destroyed). This resulted in the caller channel remaining on the system despite it having hung up. This change does the following to fix this issue: 1. The dial masquerade datastore is now removed from the caller upon dialing completion, just like the dialed targets. 2. Upon destruction of the caller all the dialed targets are also removed from the dial masquerade datastore (just in case). 3. The reference to the caller has been removed as it should not be possible for the datastore to now be valid/useful after the lifetime of the caller has ended. ASTERISK-25025 #close Change-Id: I1ef4ca5ca04980028604cc2af5d2992ac3431b3f	10 years ago
Richard Mudgett	0602409c89	chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option. Some telco switches occasionally ignore ISDN RESTART requests. The fix for ASTERISK-19608 added an escape clause for B channels in the restarting state if the telco ignores a RESTART request. If the telco fails to acknowledge the RESTART then Asterisk will assume the telco acknowledged the RESTART on the second call attempt requesting the B channel by the telco. The escape clause is good for dealing with RESTART requests in general but it does cause the next call for the restarting B channel to be rejected if the telco insists the call must go on that B channel. chan_dahdi doesn't really need to issue a RESTART request in response to receiving a cause 44 (Requested channel not available) code. Sending the RESTART in such a situation is not required (nor prohibited) by the standards. I think chan_dahdi does this for historical reasons to deal with buggy peers to get channels unstuck in a similar fashion as the chan_dahdi.conf resetinterval option. * Add the chan_dahdi.conf force_restart_unavailable_chans compatability option that when disabled will prevent chan_dahdi from trying to RESTART the channel in response to a cause 44 code. ASTERISK-25034 #close Reported by: Richard Mudgett Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65	10 years ago
Matt Jordan	561f89b693	Merge "res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback." into certified/13.1	10 years ago
Mark Michelson	c6c06bbe70	Prevent potential crash on blond transfer. Scenario: Alice calls Bob. Bob performs a blond transfer to Carol. Carol rejects the incoming call (or some other immediate circumstance causes Carol not to answer the call) What occurs in this case is that when the bridge between Alice and Bob breaks, Alice is told to masquerade into Bob's channel that had placed the call to Carol. The actual masquerade goes down without a hitch. However, a channel fixup callback that attempts to publish dial events over Stasis has a crash. The reason for this crash is that the datastore on Bob's channel that placed the outbound call to Carol only had a bare pointer to Carol's channel. Since Carol rejected the incoming call, Carol's channel has been hung up and freed, meaning accessing her channel results in a crash. The fix here is simple. The dial fixup code has been altered to hold references to the involved channels and to drop those references when freeing data. ASTERISK-25025 #close Reported by Chet Stevens Change-Id: I54eedda207b8ec7a69263353b43abe5746aea197	10 years ago
Mark Michelson	08a4cf3237	res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback. The Asterisk 13 version of the fix for outbound registration was missing a key component that set the outbound authenticator's callback that creates an authenticated request based on an old request. This was picked up by some outbound registration tests failing in the testsuite. Change-Id: I5ca9379698c606da36bc38eaffccedaf64211ce3	10 years ago
Joshua Colp	47df4e031c	res_pjsip_outbound_registration: Fix double unref on error return. When the PJSIP pjsip_regc_send function is invoked and an error status returned the caller currently decrements the reference count of the client state that it just incremented, assuming the registration callback would not have been invoked. In practice this is not correct. If the failure happens after the transaction has been set up the callback will still be invoked. This will cause the reference count to be incorrectly decremented twice, once by the registration callback and second by the caller of pjsip_regc_send. This change makes it so that whether the callback is invoked or not is known by the caller of pjsip_regc_send. Depending on this it can know whether it is responsible for decrementing the reference count of the client state or not. ASTERISK-25037 #close Reported by: Joshua Colp Change-Id: I749dc12f3a22115c49c5d7d95ff42a5fa45319de	10 years ago
Mark Michelson	11d85ea251	res_pjsip_outbound_registration: Don't fail on delayed processing: 13. This is the Asterisk 13 version of a change to master that allows for registration responses to be processed successfully potentially after the original transaction has timed out. The main difference between this and the master change is that the master version has API changes that are unacceptable for 13. For 13, this is worked around by adding a new API call that the outbound registration code uses instead. The following is the text from the master version of this commit: Odd behaviors have been observed during outbound registrations. The most common problem witnessed has been one where a request with authentication credentials cannot be created after receiving a 401 response. Other behaviors include apparently processing an incorrect SIP response. Inspecting the code led to an apparent issue with regards to how we handle transactions in outbound registration code. When a response to a REGISTER arrives, we save a pointer to the transaction and then push a task onto the registration serializer. Between the time that we save the pointer and push the task, it's possible for the transaction to be destroyed due to a timeout. It's also possible for the address to be reused by the transaction layer for a new transaction. To allow for authentication of a REGISTER request to be authenticated after the transaction has timed out, we now also hold a reference to the original REGISTER request instead of the transaction. The function for creating a request with authentication has been altered to take the original request instead of the transaction where the original request was sent. ASTERISK-25020 Reported by Mark Michelson Change-Id: If1ee5f601be839479a219424f0358a229f358f7c	10 years ago
Mark Michelson	0037ca59a6	res_pjsip_outbound_registration: Add debugging messages. When problems occur regarding outbound registrations, it currently is difficult to debug. Most off-nominal paths had warning messages, but sometimes we want to know what's going on before hitting the off-nominal path. This patch adds lots of debugging output that should give a clearer picture of what is happening with regards to outbound registrations. ASTERISK-25020 Reported by Mark Michelson Change-Id: I577bde7860be0a6c872b5bcb4d5047340bf45d45	10 years ago
Matt Jordan	f9b6336528	Merge "res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX." into certified/13.1	10 years ago
Matthew Jordan	e84fcb2464	res/res_pjsip_t38: Add missing initialization of t38faxmaxdatagram Prior to this patch, the far_max_datagram value on the UDPTL structure would remain -1 if the remote endpoint fails to provide the SDP media attribute T38FaxMaxDatagram. This can result in the INVITE request being rejected. With this patch, we will now properly initialize the value with either the default value or with the value provided by pjsip.conf's t38_udptl_maxdatagram parameter. Review: https://reviewboard.asterisk.org/r/4589 ASTERISK-24928 #close Reported by: Juergen Spies Tested by: Juergen Spies patches: pjsipT38patch20150331.txt submitted by Juergen Spies (License 6698) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 Change-Id: I15bde169fd59a224a02005fec9a439f0679a375e	10 years ago
Mark Michelson	008076ecf4	res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX. When Asterisk originates a channel to an application, the channel is hung up once the application finishes executing. When the application in question is SendFax, the Asterisk PJSIP code will attempt to reinvite the T.38 session to audio after the FAX completes. The hangup of the channel happens in the midst of this reinvite transaction. In most circumstances, this works out okay because the BYE is delayed until the reinvite transaction can complete. However, if the reinvite that Asterisk sends receives a 401/407 response, then Asterisk's attempt to re-send the reinvite with authentication will fail. This is because the session supplement in res_pjsip_t38 makes the assumption that the channel on the session will always be non-NULL. Since the channel has been hung up, though, the channel is now NULL. Attempting to operate on the channel causes a crash. This patch fixes the issue by ensuring that the channel on the session is not NULL before attempting to mess with the T.38 framehook. This patch also contains some corrections for comments that were incorrect and really confused me when I first started looking at the code. ASTERISK-25004 #close Reported by Mark Michelson Change-Id: Ic5a1230668369dda4bb13524098aed9306ab45a0	10 years ago
Joshua Colp	d7e82690f1	Merge topic 'certified-13.1' into certified/13.1 * changes: Detect potential forwarding loops based on count. More .gitignore updates .gitignore updates for master/13	10 years ago
Mark Michelson	689a251ab5	Merge "build_tools/make_version: Update version parsing for Git migration" into certified/13.1	10 years ago
Mark Michelson	c5bb6e207f	Merge "git migration: Remove support for file versions" into certified/13.1	10 years ago
Mark Michelson	6bf92934a1	Merge topic 'certified-13.1' into certified/13.1 * changes: main/editline: Add .gitignore. .gitignore: Ignore tarballs (*.gz)	10 years ago
Mark Michelson	f698b71275	Merge "Add .gitignore and .gitreview files" into certified/13.1	10 years ago
Mark Michelson	1bb6122f35	Detect potential forwarding loops based on count. A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23	10 years ago
George Joseph	cb67aae596	More .gitignore updates Added .pyc and .sha1 to the top-level .gitignore. Change-Id: I7dfc4f554d54d22947b38140d3305007503cc16a Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
George Joseph	70fab74baf	.gitignore updates for master/13 Added products of ./bootstrap Added nmenuselect and gmenuselect to menuselect/ Change-Id: Ied658463958bafc04a9aff9ebc28e40c116a6e35	10 years ago
Matt Jordan	735bea479a	build_tools/make_version: Update version parsing for Git migration External systems - such as the Asterisk Test Suite - require knowledge of the upstream branch. Unfortunately, after moving to Git, the Asterisk version currently consists of only a 'GIT" prefix followed by an object blob, e.g., GIT-as08d7. This makes it difficult for such systems to know what features are available in a particular check out of Asterisk. This patch fixes this by hardcoding the branch in a variable in the make_version script. Since the mainline branches are not changed often - typically only once a year - this is a reasonable approach to solving the problem, and is more reliable than parsing the output of 'git branch -vv'. Branches that track off of an upstream primary branch will then get the benefit of knowing which mainline branch they are currently based off of. ASTERISK-24954 #close Change-Id: I8090d5d548b6d19e917157ed530b914b7eaf9799	10 years ago
Matt Jordan	7d64479748	git migration: Remove support for file versions Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Alter the "core show file version" CLI command such that it always reports the version of Asterisk. The file version is no longer available. * main/manager: The Version key now always reports the Asterisk version. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action. - Modification of the "core show file version" CLI command. Change-Id: Ia932d3c64cd18a14a3c894109baa657ec0a85d28	10 years ago
Corey Farrell	9237e8b11e	main/editline: Add .gitignore. This patch adds a .gitignore for main/editline to ignore all build results. Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d	10 years ago
Matt Jordan	630dbcb8b4	.gitignore: Ignore tarballs (*.gz) This patch updates the root .gitignore file to ignore files with a .gz extension. This will cause git to ignore downloaded sound tarballs in the the sounds/ directory. Change-Id: I1e42fbfa02a8884231507b683e8e49ac3e278aaa	10 years ago

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