asterisk

Commit Graph

Author	SHA1	Message	Date
Florian Floimair	8539f6a657	alembic: fix suppress_q850_reason_headers column name In the original commit introducing the feature the column in the alembic script was called 'suppress_q850_reason_header'. In the code however the option is called 'suppress_q850_reason_headers' (trailing 's'). This leads to errors when ARI push configuration is used. Change-Id: Ie84808adbca6fcc9136556e4f5d741adbef5d14f	7 years ago
pk16208	84c574bb8b	chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI With tls and udp enabled asterisk generates a warning about sending message via udp instead of tls. sip notify command via cli works as expected and without warning. asterisk has to set the connection information accordingly to connection and not on presumption ASTERISK-28057 #close Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e	7 years ago
George Joseph	1843b0e2b5	app_voicemail: Remove need to subscribe to stasis app_voicemail was using the stasis cache to build and maintain a list of mailboxes that had subscribers. It then used this list to determine if a mailbox should be polled for new messages if polling was enabled. For this to work, stasis had to cache every subscription and unsubscription to the mailbox which caused a lot of overhead, both cpu and memory related. Since polling is only required when changes are being made to mailboxes outside of app_voicemail and since the number of mailboxes that don't have any subscribers is likely to be very low, all mailboxes are now polled instead of just the ones with subscribers. This paves the way for disabling the caching of stasis subscription change messages. Also fixed cleanup in some of the unit tests that not only left test users in the users list but also caused segfaults if the tests were run more than once. ASTERISK-27121 Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee	7 years ago
Joshua Colp	cd2deadb69	pjproject: Upgrade to 2.8. This change brings in PJSIP 2.8, removes all the patches that were merged upstream, and makes a minor change to support a breaking change that was done. ASTERISK-28059 Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189	7 years ago
Sean Bright	99b2e0c2ff	autoconf: Check for srtp_get_version_string() before using it Change-Id: Id2a916ff9448706090e72ff2c7fb3f5ba24a05df	7 years ago
George Joseph	926ac196af	Merge "res_srtp.c: Show linked version of libsrtp on module init" into 13	7 years ago
George Joseph	3a09d9c74c	Merge "res_pjsip: Log IPv6 addresses correctly" into 13	7 years ago
George Joseph	0b88512a4d	CI: Fix typo in testsuite git checkout Change-Id: I30024515e5b00a5044fd39fbff27d818f016b719	7 years ago
Sean Bright	44375c0616	res_srtp.c: Show linked version of libsrtp on module init Change-Id: Ib0a645d6985de5757cc4399ed2524b2d02c4f342	7 years ago
Sean Bright	d3c869c736	res_pjsip: Log IPv6 addresses correctly Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name store IPv6 addresses without enclosing brackets. This causes some log output to be confusing because it is difficult to separate the IPv6 address from a port specification. * Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6 output. * When a pj_sockaddr is not available, explicitly wrap IPv6 addresses in brackets. * When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure to also set pjsip_rx_data.pkt_info.src_addr. Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8	7 years ago
George Joseph	52324ef1a1	CI: Use proper credentials for Security testsuite checkout Can't do anonymous http checkout from Security-testsuite. Need to use same credentials as the gerrit review checkout. Change-Id: I87af68c995cb8926f5e87f9af245600d76984f05	7 years ago
Jenkins2	d4becda8ec	Merge "res_musiconhold.c: Restart MOH if previous hold just reached end-of-file" into 13	7 years ago
Jenkins2	641e5b7e63	Merge "optional_api: Remove unused nonoptreq fields" into 13	7 years ago
George Joseph	dd429cd4d8	Merge "CI: Use .gitreview to default BRANCH_NAME." into 13	7 years ago
Joshua Colp	55a306deef	Merge "res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP" into 13	7 years ago
Corey Farrell	d1f6a323a0	CI: Use .gitreview to default BRANCH_NAME. This ensures that binary modules are avoided in the master branch even if BRANCH_NAME is not set. Change-Id: I79162d2063f22fa9d6b31fde4827ace2dd5bf0da	7 years ago
Walter Doekes	d226458c5b	optional_api: Remove unused nonoptreq fields As they're not actively used, they only grow stale. The moduleinfo field itself is kept in Asterisk 13/15 for ABI compatibility. ASTERISK-28046 #close Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc	7 years ago
Sean Bright	931b68d3e6	res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP The bundled version of pjproject has a patch for Solaris compatability that changes the definition of various socket structures which we need to account for when compiling against a non-bundled version. ASTERISK-28049 #close Change-Id: Ia1ea47c433fc2d915115193ee889a752373925f0	7 years ago
Corey Farrell	436daa50ff	Build System: Resolve conflict between DESTDIR and bundled jansson. If Asterisk is built using a DESTDIR this will cause the bundled jansson to be installed to an unexpected location and we will fail to find it. Change-Id: Id033e2813261e0d45232383d44c6391122169548	7 years ago
Joshua Colp	3fd1216083	Merge "stasis_cache: Prune stasis_subscription_change messages" into 13	7 years ago
Joshua Colp	b9cf678d18	Merge "app_queue: Update realtime queuemembers after wait_a_bit(), not before" into 13	7 years ago
Frederic LE FOLL	a5f87a3a36	res_musiconhold.c: Restart MOH if previous hold just reached end-of-file On MOH activation, moh_files_readframe() is called while the current stream attached to the channel is NULL and it calls ast_moh_files_next() immediately. However, it won't call ast_moh_files_next() again if sample reading fails. The failure may occur because res_musiconhold retains the last sample reading position in the channel data and MOH during the previous hold/retrieve just reached EOF. Obviously, a bit of bad luck is required here. * Restructured moh_files_readframe() to try a second time to start MOH if there was no stream setup and the saved position was at EOF. Also added comments describing what is going on for each step. ASTERISK-28029 Change-Id: I1508cf2c094f8feca22d6f76deaa9fdfa9944860	7 years ago
lvl	eda1af091e	app_queue: Update realtime queuemembers after wait_a_bit(), not before This ensures the most up-to-date information is used for the next call attempt. ASTERISK-28032 Change-Id: I02fc17c6ffb50bb60ea97c2d2e6023e8061815ce	7 years ago
Sean Bright	b0ef54af71	res_pjproject: Add utility functions to convert between socket structures Currently, to convert from a pj_sockaddr to an ast_sockaddr, the address needs to be rendered to a string and then parsed into the correct structure. This also involves a call to getaddrinfo(3). The same is true for the inverse operation. Instead, because we know the internal structure of both ast_sockaddr and pj_sockaddr, we can translate directly between the two without the need for an intermediate string. Change-Id: If0fc4bba9643f755604c6ffbb0d7cc46020bc761	7 years ago
George Joseph	60b703d9ca	Merge "http.c: Give HTTP error response when received lines are too long." into 13	7 years ago
George Joseph	87550706c8	stasis_cache: Prune stasis_subscription_change messages The stasis cache provides a way to reconstruct the current state of topic subscribers. Unfortunately, since every subscribe and unsubscribe is cached, the cache continues to grow unabated while asterisk is running. This patch removes subscribe messages from the cache when the corresponding unsubscribe is received. This patch also registers the cache containers with ao2 so that if AO2_DEBUG is turned on, you can list the container and get its stats from the CLI. ASTERISK-27121 Change-Id: I3d18905e477f3721815da91f30da8d3fbb2d4f56	7 years ago
George Joseph	8936a264bb	Merge "app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done" into 13	7 years ago
George Joseph	5d4cf16c17	Merge "res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch" into 13	7 years ago
Rodrigo Ramírez Norambuena	17040d1ce3	app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done Change-Id: I08f88adb09f7e5813f37e70fecd787468cdb32c8	7 years ago
Chris-Savinovich	7b898dd8e1	pbx_config.c: Fix reloading module if initially declined to load Added decline if extensions.conf file not available when loading pbx_config, and also made sure everything gets properly unregistered and/or destroyed on unload. Change-Id: Ib00665106043b1be5148ffa7a477396038915854	7 years ago
Richard Mudgett	4e20fe9f06	http.c: Give HTTP error response when received lines are too long. Added a check when we receive a HTTP request line or header line that is too long. We now return an error response to the sender because we are not able to process the request. Change-Id: I6df2705435fd7dde4d5d3bdf7acec859cfb7c12d	7 years ago
Jenkins2	c0e47ee2be	Merge "make config: os-release output error." into 13	7 years ago
Joshua Colp	d0da68bfc9	Merge "res_pjsip_transport_websocket: Properly set src_name for IPv6" into 13	7 years ago
Joshua Colp	6bf0943ecb	Merge "res_fax: Handle fax gateway being started more than once." into 13	7 years ago
Richard Mudgett	f2266602a1	res_pjsip: Fix mwi_subscribe_replaces_unsolicited type mismatch ASTERISK-27988 Change-Id: Iccafdd0552ea8aaed647620fb14499f1bf341843	7 years ago
Rodrigo Ramírez Norambuena	00e5c7d0ff	make config: os-release output error. Fix not show the error "/bin/sh: /etc/os-release: No such file or directory" when the command 'make config' is run in a System without systemv. The instruction 'make config' pre execute the syntax "$(shell . /etc/os-release && echo $$ID)" to identified if system is a Slackware and Opensuse. This change prevent show the message and is send to the /dev/null Change-Id: I7f43e281a8d9405b2519fc653de82d9b8b645fdf	7 years ago
Joshua Colp	cfe8ef917d	Merge "Create --disable-binary-modules option." into 13	7 years ago
Joshua Colp	fa71f498ce	res_fax: Handle fax gateway being started more than once. The T.38 fax gateway state machine can cause the fax gateway to be started more than once on a channel depending on the responses of the remote endpoint. This would previously leak the channel name, channel unique id, and underlying fax engine state. This change instead makes it so that if the fax gateway session is already present and not reserved the fax gateway is not started again. ASTERISK-27981 Change-Id: I552d95086860cb18f2522ee40ef47b13b6da2e0e	7 years ago
George Joseph	36a03b673a	Merge "alembic: increase uri column size" into 13	7 years ago
George Joseph	7ede0ca5b3	Merge "res/res_rtp_asterisk: remove debug traces generated by an empty frame" into 13	7 years ago
Sean Bright	f3188a634d	res_pjsip_transport_websocket: Properly set src_name for IPv6 SIP responses over WebSockets when the client is using IPv6 have invalid Via headers according to RFC 3261. The 'received' header parameter should not be wrapped in brackets if it is an IPv6 address. When src_name is populated by the built-in PJSIP transports, the code uses pj_sockaddr_print() with 'flags' set to 0, meaning that the brackets are not rendered around IPv6 addresses. This may be related to ASTERISK~27101. See also: https://github.com/onsip/SIP.js/pull/594 ASTERISK-28020 #close Change-Id: I8ea9d289901b837512bee2ca2535e3dc14f04d77	7 years ago
Corey Farrell	abb04622f7	Create --disable-binary-modules option. This new option can be passed for ./configure or ./tests/CI/buildAsterisk.sh to prevent download/install of binary modules. Normally enabling the categories MENUSELECT_CODECS or MENUSELECT_RES will result in binary modules being enabled even if the build target is incompatible with those modules. This includes CI scripts which enable categories before disabling specific modules. If more binary modules are offered in the future this will help avoid accidentally downloading them if unwanted or incompatible. Adding a binary module will only require creating a new menuselect entry similar to the existing ones, it will not be necessary to modify the CI scripts. Change-Id: I6b1bd1c75a2e48f05b8b8a45b7a7a2d00a079166	7 years ago
Jenkins2	37102eb449	Merge "chan_sip: improved ip:port finding of peers for non-UDP transports." into 13	7 years ago
George Joseph	8e904d22cf	Merge "sample_configs: noload res_hep.so by default" into 13	7 years ago
Joshua Colp	c2cafddaf3	Merge "AMI: be less verbose when adding HTTP headers to AMI/HTTP messages." into 13	7 years ago
Jaco Kroon	2a5d408733	chan_sip: improved ip:port finding of peers for non-UDP transports. Also remove function peer_ipcmp_cb since it's not used (according to rmudgett). Prior to `b2c4e8660a` (ASTERISK_27457) insecure=port was the defacto standard. That commit also prevented insecure=port from being applied for sip/tcp or sip/tls. Into consideration there are three sets of behaviour: 1. "previous" - before the above commit. 2. "current" - post above commit, pre this one. 3. "new" - post this commit. The problem that the above commit tried to address was guests over TCP. It succeeded in doing that but broke transport!=udp with host!=dynamic. This commit attempts to restore sane behaviour with respect to transport!=udp for host!=dynamic whilst still retaining the guest users over tcp. It should be noted that when looking for a peer, two passes are made, the first pass doesn't have SIP_INSECURE_PORT set for the searched-for peer, thus looking for full matches (IP + Port), the second pass sets SIP_INSECURE_PORT, thus expecting matches on IP only where the matched peer allows for that (in the author's opinion: UDP with insecure=port, or any TCP based, non-dynamic host). In previous behaviour there was special handling for transport=tcp\|tls whereby a peer would match during the first pass if the utilized transport was TCP\|TLS (and the peer allowed that specific transport). This behaviour was wrong, or dubious at best. Consider two dynamic tcp peers, both registering from the same IP (NAT), in this case either peer could match for connections from an IP. It's also this behaviour that prevented SIP guests over tcp. The above referenced commit removed this behaviour, but kept applying the SIP_INSECURE_PORT only to WS\|WSS\|UDP. Since WS and WSS is also TCP based, the logic here should fall into the TCP category. This patch updates things such that the previously non-explicit (TCP behaviour) transport test gets performed explicitly (ie, matched peer must allow for the used transport), as well as the indeterministic source-port nature of the TCP protocol is taken into account. The new match algorithm now looks like: 1. As per previous behaviour, IP address is matched first. 2. Explicit filter with respect to transport protocol, previous behaviour was semi-implied in the test for TCP pure IP match - this now made explicit. 3. During first pass (without SIP_INSECURE_PORT), always match on port. 4. If doing UDP, match if matched against peer also has SIP_INSECURE_PORT, else don't match. 5. Match if not a dynamic host (for non-UDP protocols) 6. Don't match if this is WS\|WSS, or we can't trust the Contact address (presumably due to NAT) 7. Match (we have a valid Contact thus if the IP matches we have no choice, this will likely only apply to non-NAT). To logic-test this we need a few different scenarios. Towards this end, I work with a set number of peers defined in sip.conf: [peer1] host=1.1.1.1 transport=tcp [peer2] host=1.1.1.1 transport=udp [peer3] host=1.1.1.1 port=5061 insecure=port transport=udp [peer4] host=1.1.1.2 transport=udp,tcp [peer5] host=dynamic transport=udp,tcp Test cases for UDP: 1 - incoming UDP request from 1.1.1.1: - previous: - pass 1: * peer1 or peer2 if from port 5060 (indeterminate, depends on peer ordering) * peer3 if from port 5061 * peer5 if registered from 1.1.1.1 and source port matches - pass 2: * peer3 - current: as per previous. - new: - pass 1: * peer2 if from port 5060 * peer3 if from port 5061 * peer5 if registered from 1.1.1.1 and source port matches - pass 2: * peer3 2 - incoming UDP request from 1.1.1.2: - previous: - pass 1: * peer5 if registered from 1.1.1.2 and port matches * peer4 if source port is 5060 - pass 2: * no match (guest) - current: as previous. - new as previous (with the variation that if peer5 didn't have udp as allowed transport it would not match peer5 whereas previous and current code could). 3 - incoming UDP request from anywhere else: - previous: - pass 1: * peer5 if registered from that address and source port matches. - pass 2: * peer5 if insecure=port is additionally set. * no match (guest) - current - as per previous - new - as per previous Test cases for TCP based transports: 4 - incoming TCP request from 1.1.1.1 - previous: - pass 1 (indeterministic, depends on ordering of peers in memory): * peer1; or * peer5 if peer5 registered from 1.1.1.1 (irrespective of source port); or * peer2 if the source port happens to be 5060; or * peer3 if the source port happens to be 5061. - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer1 or peer2 if from source port 5060 * peer3 if from source port 5060 * peer5 if registered as 1.1.1.1 and source port matches - pass 2: * no match (guest) - new: - pass 1: * peer 1 if from port 5060 * peer 5 if registered and source port matches - pass 2: * peer 1 5 - incoming TCP request from 1.1.1.2 - previous (indeterminate, depends on ordering): - pass 1: * peer4; or * peer5 if peer5 registered from 1.1.1.2 - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer4 if source port is 5060 * peer5 if peer5 registered as 1.1.1.2 and source port matches - pass 2: * no match (guest). - new: - pass 1: * peer4 if source port is 5060 * peer5 if peer5 registered as 1.1.1.2 and source port matches - pass 2: * peer4 6 - incoming TCP request from anywhere else: - previous: - pass 1: * peer5 if registered from that address - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer5 if registered from that address and port matches. - pass 2: * no match (guest) - new: as per current. It should be noted the test cases don't make explicit mention of TLS, WS or WSS. WS and WSS previously followed UDP semantics, they will now enforce source port matching. TLS follow TCP semantics. The previous commit specifically tried to address test-case 6, but broke test-cases 4 and 5 in the process. ASTERISK-27881 #close Change-Id: I61a9804e4feba9c7224c481f7a10bf7eb7c7f2a2	7 years ago
Jaco Kroon	b528715fa0	AMI: be less verbose when adding HTTP headers to AMI/HTTP messages. All HTTP/AMI message headers are being sent to the verbose channel. There are multiple places this is happening. Consolidate the loop into a function. Drop the debug/verbose message. Convert to using ast_asprintf to perform the length calculation, memory allocation and snprintf all in one step. Change-Id: Ic45e673fde05bd544be95ad5cdbc69518207c1a1	7 years ago
Matthew Fredrickson	adb3195697	sample_configs: noload res_hep.so by default Change disables loading of res_hep.so in default installation. Loading res_hep has a performance impact whether it's used or not. This disables loading of it in sample config files. Change-Id: I5ec150cf941634fabc72973e5bf1a965cb0ef9d0 (cherry picked from commit `c8bacd45f1`)	7 years ago
Jenkins2	6c4a029bbe	Merge "chan_sip: remove unnecessary ast_channel_unlock(peer) as RAII looks after it" into 13	7 years ago
Florian Floimair	1928e00d90	alembic: increase uri column size When mobile SIP clients register with Asterisk that use some sort of push notifications, the URI can get quite lengthy due to the additional push-service annotations (things like tokens, pn-type, etc.) contained in it. ASTERISK-28022 #close Change-Id: I08e17b37af5a446c36cc7d1f903861f81fd41cca	7 years ago

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