asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	1fccedf0fd	Merge "pjproject_bundled: raise timeout value used when downloading" into 13	8 years ago
zuul	2f2811c2dc	Merge "res_xmpp: Don't crash when trying to send a message without a connection" into 13	8 years ago
zuul	b7b171c06f	Merge "res_xmpp: Include client name in connection related error messages" into 13	8 years ago
Sean Bright	d9d2beba1c	res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts chan_sip sets the hangup cause code to AST_CAUSE_REQUESTED_CHAN_UNAVAIL (44) when a channel is hung up due to an RTP timeout. So do the same when it happens with PJSIP for parity. Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8	8 years ago
zuul	68d523a1af	Merge "audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor." into 13	8 years ago
Kevin Harwell	552cf009c0	Update for 13.15.0-rc1	8 years ago
Kevin Harwell	f1b34e6eb4	AMI: Updated version Updated the AMI version for the following reason (see CHANGES for more details): The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now contains a new optional parameter, 'MatchHeader'. Change-Id: I9aeac4decc89f9b464b3f026e97c7ef1acc79242	8 years ago
Kevin Harwell	e6aeeabddf	pjproject_bundled: raise timeout value used when downloading After configuring Asterisk with '--with-pjproject-bundled' the configure/build process attempts to download pjproject from its download site. Currently, a timeout of 10 seconds is used that will stop the download process if pjproject has not been fully downloaded in that time. For some systems this was not enough time and the process was timing out too early. This patch raises the download timeout value to '60'. Also, this patch fixes another bug where the DOWNLOAD_TIMEOUT variable was not being properly exported due to a naming error. DOWNLOAD_MAX_TIMEOUT is now properly renamed to DOWNLOAD_TIMEOUT. ASTERISK-26814 #close Change-Id: Ia56e4e8a3d39db76bc8a1852b2cf07ec10b39842	8 years ago
Sean Bright	0939a19cff	res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus The documentation for JABBER_STATUS (and the deprecated JabberStatus app) indicate that a return value of 7 indicates that the specified buddy was not in the roster. It also indicates that you can specify a "bare" JID (one without a resource). Unfortunately the actual behavior does not match the documented behavior. Assuming that our roster includes the buddy online and available "valid@example.org/Valid" and does not include the buddy "invalid@example.org", the JABBER_STATUS() function returns the following before this patch: +------------------------------+------------+--------------------------+ \| Buddy \| Status \| Result \| +------------------------------+------------+--------------------------+ \| valid@example.org \| Online \| 7 (Not in roster) \| \| valid@example.org/Valid \| Online \| 1 (Online) \| \| valid@example.org/Invalid \| N/A \| 7 (Not in roster) \| \| invalid@example.org \| N/A \| Error logged, no return \| \| invalid@example.org/Valid \| N/A \| Error logged, no return \| +------------------------------+------------+--------------------------+ And after this patch: +------------------------------+------------+--------------------------+ \| Buddy \| Status \| Result \| +------------------------------+------------+--------------------------+ \| valid@example.org \| Online \| 1 (Online) \| \| valid@example.org/Valid \| Online \| 1 (Online) \| \| valid@example.org/Invalid \| N/A \| 6 (Offline) \| \| invalid@example.org \| N/A \| 7 (Not in roster) \| \| invalid@example.org/Valid \| N/A \| 7 (Not in roster) \| +------------------------------+------------+--------------------------+ This brings the behavior in line with the documentation. ASTERISK-23510 #close Reported by: Anthony Critelli Change-Id: I9c3241035363ef4a6bdc21fabfd8ffcd9ec657bf	8 years ago
Sean Bright	a487f6fb97	res_xmpp: Don't crash when trying to send a message without a connection If we never establish a connection to our Jabber server, iksemel never sets up its internal transport pointer, so attempting to send a message dereferences a NULL pointer and causes a crash. ASTERISK-21855 #close Reported by: Jeremy Kister Change-Id: I204a568894e4a53ab929783ecc594a000f04d79c	8 years ago
Sean Bright	90fb1fca41	res_xmpp: Include client name in connection related error messages ASTERISK-25622 #close Reported by: Sean Darcy Change-Id: I8472cb7bfb58d411a3cfbd482da98cae2d94d1e9	8 years ago
zuul	4fcb8d807e	Merge "CHANNEL(callid): Give dialplan access to the callid." into 13	8 years ago
zuul	b79e67ba47	Merge "res_pjsip_session: Enable RFC3578 overlap dialing support." into 13	8 years ago
zuul	5f75cc8279	Merge "res_pjsip_messaging: Check URI type before dereferencing" into 13	8 years ago
zuul	d7c52125e9	Merge "Revert "app_queue: Handle the caller being redirected out of a queue bridge"" into 13	8 years ago
Joshua Colp	f762dfaf2a	Merge "app_queue: Member stuck as pending after forwarding previous call from queue" into 13	8 years ago
Sebastian Gutierrez	e196190f11	cdr: Allow setting of user field from 'h' extension The CDR code previously did not allow the user field to be set from the 'h' extension in the dialplan. This change removes that limitation and allows it to be set. ASTERISK-26818 Change-Id: I0fed8a79b5e408bac4e30542b8f33a61c5ed9aa6	8 years ago
zuul	14a9a6fc09	Merge "pjsip: prevent memory corruption on creation of xml bodies" into 13	8 years ago
Richard Begg	398e5ec16c	res_pjsip_session: Enable RFC3578 overlap dialing support. Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6	8 years ago
zuul	d7ba743329	Merge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references." into 13	8 years ago
zuul	72ae513f15	Merge "res_hep: Capture actual transport type in use" into 13	8 years ago
Sean Bright	218f618095	res_hep: Capture actual transport type in use Rather than hard-coding UDP, allow consumers of the HEP API to specify which protocol is in use. Update the PJSIP provider to pass in the current protocol type. ASTERISK-26850 #close Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978	8 years ago
Sean Bright	1c8b81a2a4	Revert "app_queue: Handle the caller being redirected out of a queue bridge" This reverts commit `163e9e53dc`. Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b	8 years ago
Sean Bright	b3cc20799b	res_pjsip_messaging: Check URI type before dereferencing We aren't validating that the URI we just parsed is a SIP/SIPS one before trying to access the user, host, and port members of a possibly uninitialized structure. Also update the MessageSend documentation to indicate what 'from' formats are accepted. ASTERISK-26484 #close Reported by: Vinod Dharashive Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30	8 years ago
Joshua Elson	91c97b5da5	pjsip: prevent memory corruption on creation of xml bodies ASTERISK-26776 #close Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2	8 years ago
Sean Bright	7f34c11b6a	bridge_softmix: Ignore non-voice frames from translator Some codecs - codec_speex specifically - take voice frames and return other types of frames, like CNG. If we subsequently treat those as voice frames, we'll run into trouble when destroying the frame because of the requirement that each voice frame have an associated format. ASTERISK-26880 #close Reported by: Kirsty Tyerman Change-Id: I43f8450c48fb276ad8b99db8512be82949c1ca7c	8 years ago
zuul	fdea369852	Merge "res/res_pjsip_session: Only check localnet if it is defined" into 13	8 years ago
Aaron An	d5b480afca	audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor. Fixed a bug in function "ast_audiohook_write_frame" that checked the variable other_factory_samples and only flushed the factories, so they would be in sync, when other_factory_samples > 0. When there is not any rtp incoming the variable other_factory_samples will be 0, and although the result of "our_factory_ms - other_factory_ms" may be very large, this led to the record file not syncing. ASTERISK-26875 #close Reported-by: Aaron An Tested-by: Aaron An Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22	8 years ago
Joshua Colp	9613391868	Merge "thread safety: Don't use getprotobyname()" into 13	8 years ago
Sean Bright	38cebc73a3	thread safety: Don't use getprotobyname() POSIX does not require getprotobyname() to be thread safe and some implementations use static memory which causes issues when multiple threads are used. Further, our usage of it today is just to ultimately get IPPROTO_TCP for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48	8 years ago
Sean Bright	265455bc2d	res_rtp_asterisk: Pass correct data length to ast_rtcp_interpret We are currently passing in the capacity of the read buffer instead of the number of bytes that we actually read off the wire. Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36	8 years ago
Joshua Colp	baeabb82ea	Merge "app_queue: Fix locking behavior in stasis message handlers" into 13	8 years ago
Joshua Colp	1b828e50fe	Merge "chan_sip: Add rtcp-mux support" into 13	8 years ago
Joshua Colp	130af0ab80	Merge "res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped." into 13	8 years ago
Joshua Colp	a2d29a7545	Merge "res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed." into 13	8 years ago
Joshua Colp	e76947bd88	Merge "app_confbridge: Fix ConfbridgeTalking AMI event description." into 13	8 years ago
Joshua Colp	6da1b60918	Merge "res_pjsip_sdp_rtp.c: Fix cut-n-paste error" into 13	8 years ago
Robert Mordec	76afb9e18a	app_queue: Member stuck as pending after forwarding previous call from queue Queue member will get stuck in pending_members if queue calls a device that is different from the one observed for state changes. This patch removes members from pending_members as a result of channel stasis events such as blind or attended transfers and hangup. ASTERISK-26862 #close Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727	8 years ago
Richard Mudgett	60b372a883	CHANNEL(callid): Give dialplan access to the callid. * Added CHANNEL(callid) to retrieve the call identifier log tag associated with the channel. Dialplan now has access to the call log search key associated with the channel so it can be saved in case there is a problem with the call. ASTERISK-26878 Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f	8 years ago
Sean Bright	9a57b24e17	app_queue: Fix locking behavior in stasis message handlers The queue_stasis_data structure contains various mutable fields that require appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and 'caller_uniqueid' fields need to be locked when read from or written to. Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088	8 years ago
Joshua Colp	161fe61a0f	Merge "res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport" into 13	8 years ago
Sean Bright	8721d0bf1b	chan_sip: Add rtcp-mux support ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639	8 years ago
Richard Mudgett	792171ea9e	app_confbridge: Fix ConfbridgeTalking AMI event description. Thanks to Chris Howard for pointing this out on the wiki. Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705	8 years ago
Richard Mudgett	047fb7f11e	res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed. struct ast_rtcp does not define the dtls member if SRTP is not enabled. ASTERISK-26732 Change-Id: Id15ea212e04490e012f2cf4a56818b4dd948875e	8 years ago
Richard Mudgett	a75f02c089	res_pjsip_sdp_rtp.c: Fix cut-n-paste error We were inadvertenly referencing the cos_video option to determine if we should set the tos_audio and cos_audio value on the RTP instance. Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0	8 years ago
Matt Jordan	776ffd7724	res/res_pjsip_session: Only check localnet if it is defined If local_net is not defined on a transport, transport_state->localnet will be NULL. ast_apply_ha will, be default, return AST_SENSE_ALLOW in this case, causing the external_media_address, if set, to be skipped. This patch causes us to only check if we are sending within a network if local_net is defined. ASTERISK-26879 #close Change-Id: Ib661c31a954cabc9c99f1f25c9c9a5c5b82cbbfb	8 years ago
Richard Begg	139bc3495f	res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport Currently a wildcard address is used for the local RTP socket, which will not always result in the same address as used by the SIP socket (e.g. if explicit transport addresses are configured). Use the transport's host address when binding new local RTP sockets if available. ASTERISK-26851 Change-Id: I098c29c9d1f79a4f970d72ba894874ac75954f1a	8 years ago
Joshua Colp	7ea7797e12	res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped. This change removes an assumption that when DTLS is stopped an RTCP session will be present on the RTP session. This is not always the case. ASTERISK-26732 Change-Id: Ib9f7c09ce0b005efe362dbcc8795202b18f94611	8 years ago
George Joseph	9b756662a8	res_pjsip: Symmetric transports A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f	8 years ago
Joshua Colp	57be9cf8f9	Merge "Add rtcp-mux support" into 13	8 years ago

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