4010 Commits (7819f95791fe0ca0e0cdc417e2687a5900444053)

Author	SHA1	Message	Date
Richard Mudgett	5b4e2ec267	res_pjsip: Fix pointer use after unref. ... Change-Id: I4b6e1b0070563eeaee223cb58326f1b962ed5bc1	9 years ago
Richard Mudgett	6f793ac149	res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member. ... * create_rtp(): Eliminate use of deprecated transport struct member. That member and several others in the transport structure were deprecated because of an infinite loop created when using realtime configuration. See 2451d4e455 ASTERISK-26851 Change-Id: I0533aa13c9ce3c6cc394e0fd2b5bf1cd1b2ef3bc	9 years ago
Joshua Colp	270b485f04	pjsip: Add Alembic for PUBLISH support. ... This change adds database tables for the PUBLISH support so it can be configured using realtime. A minor fix to the res_pjsip_publish_asterisk module was done so that it read the sorcery configuration from the correct section. Finally the sample configuration files have been updated. ASTERISK-26928 Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952	9 years ago
zuul	d2e6eb2afe	Merge "Unused realtime MOH classes not purged on 'moh reload'"	9 years ago
zuul	a385db5333	Merge "res_pjsip_session: Allow BYE to be sent on disconnected session."	9 years ago
Richard Mudgett	f2ee8ac21e	res_pjsip_sdp_rtp.c: Don't alter global addr variable. ... * create_rtp(): Fix unexpected alteration of global address_rtp if a transport is bound to an address. * create_rtp(): Fix use of uninitialized memory if the endpoint RTP media address is invalid or the transport has an invalid address. ASTERISK-26851 Change-Id: Icde42e65164a88913cb5c2601b285eebcff397b7	9 years ago
Daniel Journo	6c3ae397cb	Unused realtime MOH classes not purged on 'moh reload' ... Purge Realtime MOH classes on 'moh reload' even when musiconhold.conf hasn't changed. ASTERISK-25974 #close Change-Id: I42c78ea76528473a656f204595956c9eedcf3246	9 years ago
Richard Mudgett	a889621b14	res_pjsip: Fix transport ref leak. ... We were leaking a transport ref in multihomed_on_rx_message() which resulted in the FRACK about excessive ref counts. ASTERISK-26916 #close Change-Id: I7a96658a9614a060565bb9ad51cb1c9c11ee145f	9 years ago
Joshua Colp	48be02c5d8	res_pjsip_session: Allow BYE to be sent on disconnected session. ... It is perfectly acceptable for a BYE to be sent on a disconnected session. This occurs when we respond to a challenge to the BYE for authentication credentials. ASTERISK-26363 Change-Id: I6ef0ddece812fea6665a1dd2549ef44fb9d90045	9 years ago
zuul	7898aad02d	Merge "res_pjsip_config_wizard: Add 2 new parameters to help with proxy config"	9 years ago
Joshua Colp	7581bb4f5f	Merge "srtp: Allow zero as tag value for a sRTP Crypto Suite."	9 years ago
zuul	410a5ac0fa	Merge "Add DTLS sanity check."	9 years ago
zuul	5a530171ca	Merge "res_musiconhold: Don't chdir() when scanning MoH files"	9 years ago
Alexander Traud	e76cc51d5e	srtp: Allow zero as tag value for a sRTP Crypto Suite. ... ASTERISK-25490 #close Change-Id: I1c5fc0942c33c96d62b24203aad0f1e1a1a0131f	9 years ago
Joshua Colp	f43cfb81d9	Merge "res_xmpp: Fix ref counting issue"	9 years ago
George Joseph	2fe52174de	res_pjsip_config_wizard: Add 2 new parameters to help with proxy config ... Two new parameters have been added to the pjsip config wizard. * Setting 'sends_line_with_registrations' to true will cause the wizard to skip the creation of an identify object to match incoming request to the endpoint and instead add the line and endpoint parameters to the outbound registration object. * Setting 'outbound_proxy' is a shortcut for adding individual endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy parameters. Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0 (cherry picked from commit a827892ff7) (cherry picked from commit 27344675be)	9 years ago
Joshua Colp	d0ada2246e	Merge "res_xmpp: Use incremental backoff when a read error occurs"	9 years ago
Richard Mudgett	3d8899bacf	Add DTLS sanity check. ... Change-Id: Ib32612cf6c7ce9213a11b9cba82f630f8cd3564b	9 years ago
Sean Bright	d22c678999	res_musiconhold: Don't chdir() when scanning MoH files ... There doesn't appear to be any reason that we are chdir'ing in moh_scan_files, and in the event of an Asterisk crash, the core files may not get written because we have changed into a read-only directory. ASTERISK-23996 #close Reported by: Walter Doekes Change-Id: Iac806dce01b3335963fbd62d4b4da9a65c614354	9 years ago
Joshua Colp	3bdf876b04	Merge "res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts"	9 years ago
Sean Bright	d5a8799c4b	res_xmpp: Use incremental backoff when a read error occurs ... If a read error occurs, we immediately attempt a reconnect without any delay. Instead, let's sleep and backoff up to 60 seconds before we try again. ASTERISK-24712 #close Reported by: Matthias Urlichs Change-Id: I6fe10ef4734837727437beab715e336777f13f48	9 years ago
zuul	f27c1c7dc2	Merge "res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus"	9 years ago
zuul	8982d0e007	Merge "res_xmpp: Include client name in connection related error messages"	9 years ago
Sean Bright	d08c69a9e2	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	9 years ago
Joshua Colp	6666deb907	Merge "res_xmpp: Don't crash when trying to send a message without a connection"	9 years ago
zuul	1a626ffb89	Merge "res_xmpp: Correctly check return value of SSL_connect"	9 years ago
Sean Bright	98a88e9ffa	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	9 years ago
Sean Bright	be94105d6d	res_xmpp: Try to provide useful errors messages from OpenSSL ... If any errors occur during the TLS connection setup, we currently dump a fairly generic error message. So instead we try to pull in something useful from OpenSSL to report instead. ASTERISK-24712 Reported by: Matthias Urlichs Change-Id: I288500991a9681f447d92913b11fedaf426087f4	9 years ago
Sean Bright	ee81ee1f14	res_xmpp: Fix ref counting issue ... The only remaining reference to the endpoint is in the endpoints container, and because it is unlinked in ast_endpoint_shutdown, we don't have to explicitly cleanup the endpoint ourselves. Change-Id: I912a2692e52d3e2ed445b32d8ae3f9004bc2f2e8	9 years ago
Sean Bright	9493981419	res_xmpp: Correctly check return value of SSL_connect ... SSL_connect returns non-zero for both success and some error conditions so simply negating is inadequate. Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1	9 years ago
Sean Bright	7657c279b5	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	9 years ago
Sean Bright	0136ec12a3	res_xmpp: Include client name in connection related error messages ... ASTERISK-25622 #close Reported by: Sean Darcy Change-Id: I8472cb7bfb58d411a3cfbd482da98cae2d94d1e9	9 years ago
Joshua Colp	c1ab8ca74c	Merge "res_pjsip_session: Enable RFC3578 overlap dialing support."	9 years ago
zuul	06c9966608	Merge "res_pjsip_messaging: Check URI type before dereferencing"	9 years ago
Richard Begg	6b7697ed48	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	9 years ago
Sean Bright	d4fcf196a2	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	9 years ago
Sean Bright	6b4b87787c	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	9 years ago
Joshua Colp	f5603cb1ec	Merge "res/res_pjsip_session: Only check localnet if it is defined"	9 years ago
Sean Bright	516e028b44	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	9 years ago
Joshua Colp	77582634d7	Merge "res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped."	9 years ago
Joshua Colp	0db211dc64	Merge "res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed."	9 years ago
Joshua Colp	7f87cd7b4e	Merge "res_pjsip_sdp_rtp.c: Fix cut-n-paste error"	9 years ago
Joshua Colp	15c72b3239	Merge "res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport"	9 years ago
Richard Mudgett	82982a191c	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	9 years ago
Joshua Colp	732367e806	Merge "res_pjsip: Symmetric transports"	9 years ago
Richard Mudgett	49b1f1ca16	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	9 years ago
Matt Jordan	e6dc28b78f	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	9 years ago
Joshua Colp	76e64f5589	Merge "RFC sdp: Initial SDP creation"	9 years ago
Richard Begg	44568fc712	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	9 years ago
George Joseph	5013d8f5d3	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	9 years ago