asterisk

Commit Graph

Author	SHA1	Message	Date
Corey Farrell	8971a7ff3c	Loader: Remove unneeded load_pri declarations. Instead of specifying AST_MODFLAG_LOAD_ORDER with load_pri AST_MODPRI_DEFAULT just use AST_MODFLAG_DEFAULT. Change-Id: I0123258eafce324249433a69df15a85cc16e509f	8 years ago
Joshua Colp	c2dddb001a	pjsip / hep: Provide correct local address for Websockets. Previously for PJSIP the local address of WebSocket connections was set to the remote address. For logging purposes this is not particularly useful. The WebSocket API has been extended to allow the local address to be queried and this is used in PJSIP to set the local address to the correct value. The PJSIP HEP support has also been tweaked so that reliable transports always use the local address on the transport and do not try to (wrongly) guess. As they are connection based it is impossible for the source to be anything else. ASTERISK-26758 ASTERISK-27363 Change-Id: Icd305fd038ad755e2682ab2786e381f6bf29e8ca	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	9 years ago
Matt Jordan	eabb398d71	res_hep_{pjsip\|rtcp}: Decline module loads if res_hep had not loaded A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not loaded and does not have a configuration file. Previously when this occurred, checks were put in to see if the configuration was loaded successfully. While this is a good idea - and has been added to the offending function in res_hep - the reality is res_hep_pjsip and res_hep_rtcp have no business running if res_hep isn't also running. As such, this patch also adds a function to res_hep that returns whether or not it successfully loaded. Oddly enough, ast_module_check returns "everything is peachy" even if a module declined its load - so it cannot be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this function to see if they should continue to load; if it fails, they decline their load as well. ASTERISK-26096 #close Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea	10 years ago
Matt Jordan	f91a7dc993	res/res_hep_pjsip: Fix reported local IP address when bound to 'any' When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its local address the 'any' address, as opposed to the IP address we actually received the packet on. This can cause some confusion in Homer, as it will dutifully report what we send it. This patch uses the PJSIP inspection routines to determine which IP address we probably received the packet on based on the remote party's IP address. In the event that this fails, it falls back to the IP address natively reported by the transport. Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3	10 years ago
Matt Jordan	89ae4466ea	res_hep: Provide an option to pick the UUID type At one point in time, it seemed like a good idea to use the Asterisk channel name as the HEP correlation UUID. In particular, it felt like this would be a useful identifier to tie PJSIP messages and RTCP messages together, along with whatever other data we may eventually send to Homer. This also had the benefit of keeping the correlation UUID channel technology agnostic. In practice, it isn't as useful as hoped, for two reasons: 1) The first INVITE request received doesn't have a channel. As a result, there is always an 'odd message out', leading it to be potentially uncorrelated in Homer. 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. This causes RTCP information to be uncorrelated to the SIP message traffic seen by those capture nodes. In order to support both (in case someone is trying to use res_hep_rtcp with a non-PJSIP channel), this patch adds a new option, uuid_type, with two valid values - 'call-id' and 'channel'. The uuid_type option is used by a module to determine the preferred UUID type. When available, that source of a correlation UUID is used; when not, the more readily available source is used. For res_hep_pjsip: - uuid_type = call-id: the module uses the SIP Call-ID header value - uuid_type = channel: the module uses the channel name if available, falling back to SIP Call-ID if not For res_hep_rtcp: - uuid_type = call-id: the module uses the SIP Call-ID header if the channel type is PJSIP and we have a channel, falling back to the Stasis event provided channel name if not - uuid_type = channel: the module uses the channel name ASTERISK-25352 #close Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c	10 years ago
Kinsey Moore	289830cdc6	PJSIP: Enforce module load dependencies This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub have loaded properly before attempting to load any modules that depend on them since the module loader system is not currently capable of resolving module dependencies on its own. ASTERISK-24312 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/4062/ ........ Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	69032d62aa	res_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible (although it shouldn't occur) for pjproject to pass up an rdata object with a NULL msg in the msg_info. Needless to say, things that attempt to dereference this are in for a rough ride. In particular, this caused crashes in three different locations, all of which are 'low level' enough to intercept an rdata object early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3) res_pjsip/distributor Anything that can intercept an rdata object before res_pjsip/distributor should be defensive when looking at the received packet. #SIPit31 ASTERISK-24369 #close Reported by: Matt Jordan ........ Merged revisions 424618 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cc7853f40f	res_hep: Remove disabling of modules These modules were originally specified as being disabled, as they were introduced midstream in Asterisk 12. That makes it nicer for folks who are upgrading to a new release in the middle of Asterisk 12. That's not the case for Asterisk 13: it's a brand new release. There's no reason to have the modules disabled by default in that case. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420742 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0c1342bd2b	res_hep_pjsip: Use the channel name instead of the call ID when it is available During discussions with Alexandr Dubovikov at Kamailio World, it became apparent that while the SIP call ID is a useful identifier prior to an Asterisk channel being created, it is far more preferable to use the channel name (or some channel based identifier) when the channel is available. Homer is smart enough to tie the various messages together. This patch opts to use the channel name when it is available, falling back to the call ID otherwise. ........ Merged revisions 412088 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ef0c9fe4d8	res_hep/res_hep_pjsip: Add a HEPv3 capture agent module and a logger for PJSIP This patch adds the following: (1) A new module, res_hep, which implements a generic packet capture agent for the Homer Encapsulation Protocol (HEP) version 3. Note that this code is based on a patch provided by Alexandr Dubovikov; I basically just wrapped it up, added configuration via the configuration framework, and threw in a taskprocessor. (2) A new module, res_hep_pjsip, which forwards all SIP message traffic that passes through the res_pjsip stack over to res_hep for encapsulation and transmission to a HEPv3 capture server. Much thanks to Alexandr for his Asterisk patch for this code and for a lot of patience waiting for me to port it to 12/trunk. Due to some dithering on my part, this has taken the better part of a year to port forward (I still blame CDRs for the delay). ASTERISK-23557 #close Review: https://reviewboard.asterisk.org/r/3207/ ........ Merged revisions 411534 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

12 Commits (d7f5421ea42ce18490e36f61e34893b96dc5708e)