2946 Commits (63196a825673a06f84df90314e48be5aa3f67d06)

Author	SHA1	Message	Date
Corey Farrell	63196a8256	res_pjsip_dlg_options: Fix MODULEINFO section. ... Removed the extra space before "MODULEINFO" in res_pjsip_dlg_options. This extra space prevented any of the dependencies from being seen by menuselect, so building with default options would fail if PJSIP was not installed. This also makes the tool that extracts information for menuselect tolerant of multiple spaces in the future. ASTERISK-25033 #close Reported by: Peter Whisker Change-Id: Iccd54846f70c4a7a50cb5bf70b7bb5cb4bab3698	11 years ago
Matt Jordan	ec0f80b6e8	Merge "res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback." into 13	11 years ago
Mark Michelson	4b8cddfb36	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	11 years ago
Joshua Colp	415a0d0745	res_ari_device_states: Fix dependency on res_stasis_device_state. ... The res_ari_device_states module depends on res_stasis_device_state, not res_stasis_device_states. Change-Id: I26e02ad37f9e36bcc859867e2fad1b90452ec3de	11 years ago
Matt Jordan	d1bc86fc99	Merge "res_pjsip_outbound_registration: Add virtual line support." into 13	11 years ago
Joshua Colp	e332c7ed5e	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	11 years ago
Matt Jordan	37a193da18	Merge "ARI: Fix missing dependencies." into 13	11 years ago
Kevin Harwell	3fb6daeb55	res_fax: allow 2400 transmission rate according to v.27ter standard ... A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits per second. This reverts all or some of those patches since according to the v.27ter standard a rate of 2400 bits per second is also supported. One of the original patches also added 9600 bits per second support for v.27. This patch also removes that since v.27ter only supports 2400/4800 bits per second. Also, since Asterisk specifically supports v.27ter the enum was renamed to better reflect this. ASTERISK-24955 #close Reported by: Matt Jordan Change-Id: I4b9dfb6bf7eff08463ab47ee1a74224f27cae733	11 years ago
Joshua Colp	49ef81c15c	res_sorcery_config: Fix build issue due to syntax error. ... Change-Id: Ic8322f04e37842848ad72cf2871bd0378f67c4ac	11 years ago
Ashley Sanders	3278fe5327	chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR ... Sections Exist in pjsip.conf This patch modifies the current loading strategy of the pjsip configuration. If duplicate sections (e.g. sections containing the same [id/type]) are defined in [pjsip.conf], the loader will consider the configuration for the given type as invalid when the duplicate section is encountered. The entire configuration (including what was previously loaded) for the duplicate [id/type] sections will be rejected and destroyed, an error message is logged and the load processing for the given stops. ASTERISK-24996 Reported By: Ashley Sanders Change-Id: I35090ca4cd40f1f34881dfe701a329145c347aef	11 years ago
Joshua Colp	89f6719f7a	res_pjsip_outbound_registration: Add virtual line support. ... Virtual line support establishes a relationship between messages related to an outbound registration and a local endpoint. This is accomplished by attaching a parameter to the Contact of the outbound registration and looking for it on any received requests. If the parameter exists and can be matched to an outbound registration the configured endpoint is associated with the request. ASTERISK-24949 #close Reported by: Joshua Colp Change-Id: I7df909d2625479110a83fdd354c21ac539e8615d	11 years ago
Corey Farrell	d61f03c4f9	ARI: Fix missing dependencies. ... ARI modules that are generated by 'make ari-stubs' are all dependent on res_ari_model. Additionally some of the same modules depend on one or more res_stasis_* modules. ASTERISK-25027 #close Reported by: Corey Farrell Change-Id: I8e07fe7e81fedacb87232f2b6f8b5f47927b4153	11 years ago
Corey Farrell	3e4624ad21	res_pjsip: Remove incorrect MODULEINFO from presence_xml.c. ... Remove incorrect MODULEINFO block and unneeded header includes from presence_xml.c. ASTERISK-25027 Reported by: Corey Farrell Change-Id: I977c609ab9d1fe05373027c4138900f6985990eb	11 years ago
Joshua Colp	df23c8a86b	res_pjsip_outbound_registration: Fix build due to removal of transaction. ... Change-Id: I7a8a7beec3334cec304943f2dd7597eabe2e3150	11 years ago
Joshua Colp	95ab9fdb1a	Merge "res_pjsip_outbound_registration: Add debugging messages." into 13	11 years ago
Joshua Colp	0e70dc0dc8	Merge "res_pjsip_outbound_registration: Don't fail on delayed processing: 13." into 13	11 years ago
Mark Michelson	e39bd6ba46	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	11 years ago
Mark Michelson	1bf008fc76	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	11 years ago
Steve Davies	0b6410c4f8	res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS ... ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created. The resources are linked into a table, but the original alloc refs are never released. ast_strdup leak in rtp_engine.c. If ast_rtp_dtls_cfg_copy() is called twice on the same destination struct, a pointer to an alloc'd string is overwritten before the string is free'd. ASTERISK-25022 Reported by: one47 Change-Id: I62a8ceb8679709f6c3769136dc6aa9a68202ff9b	11 years ago
George Joseph	99fb87ae13	res_pjsip: Fix SEGV on pending-qualify contacts ... Permanent contacts that hadn't been qualified yet were missing their contact_status entries causing SEGVs when running CLI commands. This patch makes sure that contact_statuses are created for both dynamic and permanent contacts when they are created. It also adds checks in the CLI code to make sure there's a contact_status, just in case. ASTERISK-25018 #close Reported-by: Ivan Poddubny Tested-by: Ivan Poddubny Tested-by: George Joseph Change-Id: I3cc13e5cedcafb24c400368b515b02d7fb81e029	11 years ago
Matt Jordan	2d277996b7	Merge "Clang: Fix some more tautological-compare warnings." into 13	11 years ago
Mark Michelson	1da9ec969d	res_pjsip_outbound_authenticator: Increase CSeq on authed requests. ... The way PJSIP generates an authenticated request is to use a previous request as a template. This means that the authenticated request will have the same Call-ID, From header (including tag), and CSeq as the original request. PJSIP generates a new branch on the Via header to indicate that this is a new transaction, though. There are some SIP implementations, though, that do not notice the change in the branch and therefore will match the authed request to the original request's transaction. Since the CSeq is the same, the server will repeat the response it sent to the original request. This patch aids interoperability by increasing the CSeq of the authed request by one. ASTERISK-24845 #close Reported by: Carl Fortin Tested by: Carl Fortin Change-Id: I39c4ca52e688a9f83bcc1878371334becdc5be01	11 years ago
Matt Jordan	bf3d9db4a6	Merge "res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX." into 13	11 years ago
Mark Michelson	b3cd5bc77f	Merge "Clang: change previous tautological-compare fixes." into 13	11 years ago
Mark Michelson	eabf3b5a3c	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	11 years ago
George Joseph	f70d21b2cf	res_pjsip: Validate that contact uris start with sip: or sips: ... Currently we use pjsip_parse_hdr to validate contact uris but it appears that it allows uris without a scheme if there's a port supplied. I.E myexample.com will fail but myexample.com:5060 will pass even though it has no scheme. This causes SEGVs later on whenever the uri is used. To prevent this, permanent_contact_validate has been updated to check that the scheme is either 'sip' or 'sips'. 2 uses of possibly-null endpoint have also been fixed in create_out_of_dialog_request. ASTERISK-24999 Change-Id: Ifc17d16a4923e1045d37fe51e43bbe29fa556ca2 Reported-by: Brad Latus	11 years ago
Diederik de Groot	1bb16bedc7	Clang: change previous tautological-compare fixes. ... clang can warn about a so called tautological-compare, when it finds comparisons which are logically always true, and are therefor deemed unnecessary. Exanple: unsigned int x = 4; if (x > 0) // x is always going to be bigger than 0 Enum Case: Each enumeration is its own type. Enums are an integer type but they do not have to be *signed*. C leaves it up to the compiler as an implementation option what to consider the integer type of a particu- lar enumeration is. Gcc treats an enum without negative values as an int while clang treats this enum as an unsigned int. rmudgett & mmichelson: cast the enum to (unsigned int) in assert. The cast does have an effect. For gcc, which seems to treat all enums as int, the cast to unsigned int will eliminate the possibility of negative values being allowed. For clang, which seems to treat enums without any negative members as unsigned int, the cast will have no effect. If for some reason in the future a negative value is ever added to the enum the assert will still catch the negative value. ASTERISK-24917 Change-Id: I0557ae0154a0b7de68883848a609309cdf0aee6a	11 years ago
George Joseph	1474bb05f6	res_corosync: Add check for config file before calling corosync apis ... On some systems, res_corosync isn't compatible with the installed version of corosync so corosync_cfg_initialize fails, load_module returns LOAD_FAILURE, and Asterisk terminates. The work around has been to remember to add res_corosync as a noload in modules.conf. A better solution though is to have res_corosync check for its config file before attempting to call corosync apis and return LOAD_DECLINE if there's no config file. This lets Asterisk loading continue. If you have a res_corosync.conf file and res_corosync fails, you get the same behavior as today and the fatal error tells you something is wrong with the install. ASTERISK-24998 Change-Id: Iaf94a9431a4922ec4ec994003f02135acfdd3889	11 years ago
Diederik de Groot	d6dfc85666	Clang: Fix some more tautological-compare warnings. ... clang can warn about a so called tautological-compare, when it finds comparisons which are logically always true, and are therefor deemed unnecessary. Exanple: unsigned int x = 4; if (x > 0) // x is always going to be bigger than 0 Enum Case: Each enumeration is its own type. Enums are an integer type but they do not have to be *signed*. C leaves it up to the compiler as an implementation option what to consider the integer type of a particu- lar enumeration is. Gcc treats an enum without negative values as an int while clang treats this enum as an unsigned int. rmudgett & mmichelson: cast the enum to (unsigned int) in assert. The cast does have an effect. For gcc, which seems to treat all enums as int, the cast to unsigned int will eliminate the possibility of negative values being allowed. For clang, which seems to treat enums without any negative members as unsigned int, the cast will have no effect. If for some reason in the future a negative value is ever added to the enum the assert will still catch the negative value. ASTERISK-24917 Change-Id: Ief23ef68916192b9b72dabe702b543ecfeca0b62	11 years ago
Joshua Colp	7b57116833	res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers. ... Currently the res_pjsip_mwi module only sends an unsolicited MWI NOTIFY upon a mailbox state change (such as a new message being left, or one being deleted). In practice this is not sufficient to keep clients aware of the current MWI status. This change makes the module send unsolicited MWI NOTIFY on startup so that clients are guaranteed to have the most up to date MWI information. It also makes clients receive an unsolicited MWI NOTIFY upon registration so if they are unaware of the current MWI status they receive it. ASTERISK-24982 #close Reported by: Joshua Colp Change-Id: I043f20230227e91218f18a82c7d5bb2aa62b1d58	11 years ago
Joshua Colp	4423d5f755	Merge "res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs." into 13	11 years ago
Mark Michelson	3327560cb2	res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs. ... When SUBSCRIBE dialogs were established, we never associated the endpoint that created the subscription with the dialog we end up creating. In most cases, this ended up not causing any problems. The actual bug that was observed was that when a device that was behind NAT established a subscription with Asterisk, Asterisk would end up sending in-dialog NOTIFY requests to the device's private IP addres instead of the public address of the NAT router. When Asterisk receives the initial SUBSCRIBE from the device, res_pjsip_nat rewrites the contact to the public address on which the SUBSCRIBE was received. This allows for the dialog to have its target address set to the proper public address. Asterisk then would send a 200 OK response to the SUBSCRIBE, then a NOTIFY with the initial subscription state. The device would then send a 200 OK response to Asterisk's NOTIFY. Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat did not rewrite the address in the Contact header. Then, when the PJSIP dialog layer processed the 200 OK, PJSIP would perform a comparison between the IP address in the Contact header and its saved target address for the dialog. Since they differed, PJSIP would update the target dialog address to be the address in the Contact header. From this point, if Asterisk needed to send a NOTIFY to the device, the result was that the NOTIFY would be sent to the private address that the device placed in the Contact header. The reason why res_pjsip_nat did not rewrite the address when it received the 200 OK response was that it could not associate the incoming response with a configured endpoint. This is because on a response, the only way to associate the response to an endpoint is by finding the dialog that the response is associated with and then finding the endpoint that is associated with that dialog. We do not perform endpoint lookups on responses. res_pjsip_pubsub skipped the step of associating the endpoint with the dialog we created, so res_pjsip_nat could not find the associated endpoint and therefore couldn't rewrite the contact. This commit message is like 50x longer than the actual fix. ASTERISK 24981 #close Reported by Mark Michelson Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd	11 years ago
Joshua Colp	96e18453f4	Merge "pjsip_options: Fix non-qualified contacts showing as unavailable" into 13	11 years ago
George Joseph	b74b2cdcda	pjsip_options: Fix format specifier for int64_t rtt. ... Contact status rtt is an int64_t and needs the PRId64 macro to properly create the format specifier on 32-bit systems. Change-Id: I4b8ab958fc1e9a179556a9b4ffa49673ba9fdec7	11 years ago
George Joseph	63169e00ff	pjsip_options: Fix non-qualified contacts showing as unavailable ... The "Add qualify_timeout processing and eventing" patch introduced an issue where contacts that had qualify_frequency set to 0 were showing Unavailable instead Unknown. This patch checks for qualify_frequency=0 and create an "Unknown" contact_status with an RTT = 0. Previously, the lack of contact_status implied Unknown but since we're now changing endpoint state based on contact_status, I've had to add new UNKNOWN status so that changes could trigger the appropriate contact_status observers. ASTERISK-24977: #close Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7	11 years ago
Richard Mudgett	82bc0fd3ad	res_fax: Fix latent bug exposed by ASTERISK-24841 changes. ... Three fax related tests started failing as a result of changes made for ASTERISK-24841: tests/fax/pjsip/gateway_t38_g711 tests/fax/sip/gateway_mix1 tests/fax/sip/gateway_mix3 Historically, ast_channel_make_compatible() did nothing if the channels were already "compatible" even if they had a sub-optimal translation path already setup. With the changes from ASTERISK-24841 this is no longer true in order to allow the best translation paths to always be picked. In res_fax.c:fax_gateway_framehook() code manually setup the channels to go through slin and then called ast_channel_make_compatible(). With the previous version of ast_channel_make_compatible() this was always a no-operation. * Remove call to ast_channel_make_compatible() in fax_gateway_framehook() that now undoes what was just setup when the framehook is attached. * Fixed locking around saving the channel formats in fax_gateway_framehook() to ensure that the formats that are saved are consistent. * Fix copy pasta errors in fax_gateway_framehook() that confuses read and write when dealing with saved channel formats. ASTERISK-24841 Reported by: Matt Jordan Change-Id: I6fda0877104a370af586a5e8cf9e161a484da78d	11 years ago
Matt Jordan	e05b076827	Merge "Detect potential forwarding loops based on count." into 13	11 years ago
Mark Michelson	4f1a8dbe92	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	11 years ago
George Joseph	674b18bdf0	pjsip_options: Add qualify_timeout processing and eventing ... This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html The basic issues are that changes in contact status don't cause events to be emitted for the associated endpoint. Only dynamic contact add/delete actions update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds which is a long time. This patch makes use of the new transaction timeout feature in r4585 and provides the following capabilities... 1. A new aor/contact variable 'qualify_timeout' has been added that allows the user to specify the maximum time in milliseconds to wait for a response to an OPTIONS message. The default is 3000ms. When the timer expires, the contact is marked unavailable. 2. Contact status changes are now propagated up to the endpoint as follows... When any contact is 'Available', the endpoint is marked as 'Reachable'. When all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The existing endpoint events are generated appropriately. ASTERISK-24863 #close Change-Id: Id0ce0528e58014da1324856ea537e7765466044a Tested-by: Dmitriy Serov Tested-by: George Joseph <george.joseph@fairview5.com>	11 years ago
Matt Jordan	f1abf51b73	Merge "res_pjsip: Refactor endpt_send_request to include transaction timeout" into 13	11 years ago
Matt Jordan	ab5b38e434	Merge "res_pjsip: Add global option to limit the maximum time for initial qualifies" into 13	11 years ago
Scott Griepentrog	8d4ce7cc2b	res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced ... This change makes the send_notify of the sub_tree not happen when the sub_tree has been deleted due to the notify call failing, which avoids a crash. ASTERISK-24970 #close Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf	11 years ago
George Joseph	bf46799f0e	res_pjsip: Refactor endpt_send_request to include transaction timeout ... This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html Since we currently have no control over pjproject transaction timeout, this patch pulls the pjsip_endpt_send_request function out of pjproject and into res_pjsip/endpt_send_transaction in order to implement that capability. Now when the transaction is initiated, we also schedule our own pj_timer with our own desired timeout. If the transaction completes before either timeout, pjproject cancels its timer, and calls our tsx callback where we cancel our timer and run the app callback. If the pjproject timer times out first, pjproject calls our tsx callback where we cancel our timer and run the app callback. If our timer times out first, we terminate the transaction which causes pjproject to cancel its timer and call our tsx callback where we run the app callback. Regardless of the scenario, pjproject is calling the tsx callback inside the group_lock and there are checks in the callback to make sure it doesn't run twice. As part of this patch ast_sip_send_out_of_dialog_request was created to replace its similarly named private function. It takes a new timeout argument in milliseconds (<= 0 to disable the timeout). ASTERISK-24863 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747	11 years ago
George Joseph	1b6f6ff841	res_pjsip: Add global option to limit the maximum time for initial qualifies ... Currently when Asterisk starts initial qualifies of contacts are spread out randomly between 0 and qualify_timeout to prevent network and system overload. If a contact's qualify_frequency is 5 minutes however, that contact may be unavailable to accept calls for the entire 5 minutes after startup. So while staggering the initial qualifies is a good idea, basing the time on qualify_timeout could leave contacts unavailable for too long. This patch adds a new global parameter "max_initial_qualify_time" that sets the maximum time for the initial qualifies. This way you could make sure that all your contacts are initialy, randomly qualified within say 30 seconds but still have the contact's ongoing qualifies at a 5 minute interval. If max_initial_qualify_time is > 0, the formula is initial_interval = min(max_initial_interval, qualify_timeout * random(). If not set, qualify_timeout is used. The default is "0" (disabled). ASTERISK-24863 #close Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 Tested-by: George Joseph <george.joseph@fairview5.com>	11 years ago
Corey Farrell	0e4b997cd7	res_monitor: Add dependency on func_periodic_hook. ... OPTIONAL_API has conditionals to define AST_OPTIONAL_API and AST_OPTIONAL_API_ATTR differently based on if AST_API_MODULE is defined. Unfortunately this is inside the include protection block, so only the first status of AST_API_MODULE is respected. For example res_monitor is an optional API provider, but uses func_periodic_hook. This makes func_periodic_hook non-optional to res_monitor. ASTERISK-17608 #close Reported by: Warren Selby Change-Id: I8fcf2a5e7b481893e17484ecde4f172c9ffb5679	11 years ago
George Joseph	555b5f5d30	Add .gitignore and .gitreview files ... Add the .gitignore and .gitreview files to the asterisk repo. NB: You can add local ignores to the .git/info/exclude file without having to do a commit. Common ignore patterns are in the top-level .gitignore file. Subdirectory-specific ignore patterns are in their own .gitignore files. Change-Id: I4c8af3b8e3739957db545f7368ac53f38e99f696 Tested-by: George Joseph	11 years ago
Matthew Jordan	4cf7d0bf01	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	11 years ago
Richard Mudgett	13cd99682d	chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices. ... With this patch, chan_pjsip/res_pjsip now sets the native formats to the codecs negotiated by a call. * The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native formats to include all the negotiated audio codecs instead of only the initial preferred audio codec and later the currently received audio codec. * The audio frame handling in channel.c:ast_read() is more streamlined and will automatically adjust to changes in received frame formats. The new policy is to remove translation and pass the new frame format to the receiver except if the translation was to a signed linear format. A more long winded version is commented in ast_read() along with some caveats. * The audio frame handling in channel.c:ast_write() is more streamlined and will automatically adjust any needed translation to changes in the frame formats sent. Frame formats sent can change for many reasons such as a recording is being played back or the bridged peer changed the format it sends. Since it is a normal expectation that sent formats can change, the codec mismatch warning message is demoted to a debug message. * Removed the short circuit check in channel.c:ast_channel_make_compatible_helper(). Two party bridges need to make channels compatible with each other. However, transfers and moving channels among bridges can result in otherwise compatible channels having sub-optimal translation paths if the make compatible check is short circuited. A result of forcing the reevaluation of channel compatibility is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc options take effect consistently now. It is unfortunate that these two options are enabled by default and negate some of the benefits to the changes in channel.c:ast_read() by forcing translation through signed linear on a two party bridge. * Improved the softmix bridge technology to better control the translation of frames to the bridge. All of the incoming translation is now normally handled by ast_read() instead of splitting any translation steps between ast_read() and the slin factory. If any frame comes in with an unexpected format then the translation path in ast_read() is updated for the next frame and the slin factory handles the current frame translation. This is the final patch in a series of patches aimed at improving translation path choices. The other patches are on the following reviews: https://reviewboard.asterisk.org/r/4600/ https://reviewboard.asterisk.org/r/4605/ ASTERISK-24841 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4609/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	88b0fa7755	res_pjsip: Add an 'auto' option for DTMF Mode ... This patch adds support for automatically detecting the type of DTMF that a PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto', the channel created for an endpoint will attempt to determine if RFC 4733 DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type for the channel will be set to inband. Review: https://reviewboard.asterisk.org/r/4438 ASTERISK-24706 #close Reported by: yaron nahum patches: yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	16afee4651	res_pjsip_config_wizard: Cleanup load unload ... While investigating other unload issues I realized that the load/unload process for the config wizard was pretty ugly so I've refactored it as follows... When the res_pjsip sorcery instance is created the config_wizard bumps it's own module reference to prevent it from unloading while the sorcery instance is still active. When res_pjsip unloads and it's sorcery instance is destroyed, the config wizard unrefs itself which then allows itself to unload cleanly. Since the config wizard now can't load after res_pjsip or unload before it (which should have been the correct behavior all along), I was able to remove the chunks of code in both load_module and unload_module that handled that case. Ran the testsuite tests to insure there were no functional changes and REF_DEBUG to insure that Asterisk was shutting down cleanly with no FRACKs or leaks. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4610/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago