asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin Harwell	93ac45d3bd	res_pjsip: Add option to force G.726 to be treated as AAL2 packed. Some phones send g.726 audio packed for AAL2, which differs from what is recommended by RFC 3351. If Asterisk receives audio formatted as such when negotiating g.726 then it sounds a bit distorted. Added an option to res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 AAL2 packed. ASTERISK-25158 #close Reported by: Steve Pitts Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615	11 years ago
Alexander Traud	8f3f414d8c	tcptls: Enable multiple TLS certificate chains (RSA+ECC+DSA) for server socket. When a client connects to a server via SSL/TLS, the server commonly utilizes an RSA key-pair. However, other such algorithms exist (i.e. DSA and ECDSA), and if the server socket is configured with a certificate for either one of those, it would lose its compatibility with RSA-only clients. Now, the server socket can be configured with up to one RSA, ECDSA and DSA key each. For example, if a client is not compatible with SHA-2 hashed certificates like Nokia mobile phones, the server socket still can use RSA/SHA-1 for legacy clients and ECDSA/SHA-2 for everyone else. ASTERISK-24815 #close Reported by: Alexander Traud patches: tls_rsa_ecc_dsa.patch uploaded by Alexander Traud (License 6520) Change-Id: Iada5e00d326db5ef86e0af7069b4dfa1b979da9a	11 years ago
Corey Farrell	6b208d8c3b	Sample Configs: Fix syntax error in pjsip.conf The sample pjsip.conf has a few comment lines that are missing the semicolons at the start of the comment, causing the config to fail load. Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352	11 years ago
George Joseph	c6ed681638	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
George Joseph	51886c68dc	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
Richard Mudgett	4c2fc5b811	chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages. Incoming PJSIP call legs that have not been answered yet send unnecessary "180 Ringing" or "183 Progress" messages every time a connected line update happens. If the outgoing channel is also PJSIP then the incoming channel will always send a "180 Ringing" or "183 Progress" message when the outgoing channel sends the INVITE. Consequences of these unnecessary messages: * The caller can start hearing ringback before the far end even gets the call. * Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of the first Asterisk box. When connected line first went into Asterisk in v1.8, chan_sip received an undocumented option "rpid_immediate" that defaults to disabled. When enabled, the option immediately passes connected line update information to the caller in "180 Ringing" or "183 Progress" messages as described above. * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or "183 Progress" messages. The default is "no" to disable sending the unnecessary messages. ASTERISK-24781 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4473/ ........ Merged revisions 433338 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	aef7278af6	res_pjsip: Allow configuration of endpoint identifier query order This patch fixes previously reverted code that caused binary incompatibility problems with some modules. And like the original patch it makes sure that no matter what order the endpoint identifier modules were loaded, priority is given based on the ones specified in the new global 'endpoint_identifier_order' option. ASTERISK-24840 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4489/ ........ Merged revisions 433028 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	d42c6adb1a	Revert - res_pjsip: Allow configuration of endpoint identifier query order Due to a break in binary compatibility with some other modules these changes are being reverted until the issue can be resolved. ASTERISK-24840 Reported by: Mark Michelson ........ Merged revisions 432868 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	1ce529d30e	res_pjsip: allow configuration of endpoint identifier query order It's possible to have a scenario that will create a conflict between endpoint identifiers. For instance an incoming call could be identified by two different endpoint identifiers and the one chosen depended upon which identifier module loaded first. This of course causes problems when, for example, the incoming call is expected to be identified by username, but instead is identified by ip. This patch adds a new 'global' option to res_pjsip called 'endpoint_identifier_order'. It is a comma separated list of endpoint identifier names that specifies the order by which identifiers are processed and checked. ASTERISK-24840 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4455/ ........ Merged revisions 432638 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	55eb8fc068	pjsip: Remove "contact" type from pjsip.conf.sample The "contact" object is not meant to be configured from the pjsip.conf configuration file. It is meant to be created as a result of a registration and stored elsewhere. ASTERISK-24085 #close Reported by: Rusty Newton ........ Merged revisions 431860 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	023fa0f9e8	Add support for the ca_list_path option for PJSIP transports. This allows for a path to be specified that has a collection of CA certificates in it. ASTERISK-24575 #close Reported by cloos Patches: pj-ca-path-trunk.diff uploaded by cloos (License #5956) Review: https://reviewboard.asterisk.org/r/4344 ........ Merged revisions 430709 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	7f8b7ace72	res_pjsip_sdp_rtp: Add support for optimistic SRTP. Optimistic SRTP is the ability to enable SRTP but not have it be a fatal requirement. If SRTP can be used it will be, if not it won't be. This gives you a better chance of using it without having your sessions fail when it can't be. Encrypt all the things! Review: https://reviewboard.asterisk.org/r/3992/ ........ Merged revisions 428222 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	69f29e627f	Make the disable_tcp_switch PJSIP system object enabled by default. Testing has shown repeatedly that PJSIP's default behavior of switching automatically to TCP for large messages can cause issues. The most common issues are that devices that we are communicating with do not handle the switch to TCP gracefully, thus causing situations such as broken calls or broken subscriptions. Now, in order to have this behavior happen, you must opt into it. The sample file has been updated to warn that enabling the TCP switch behavior may cause issues for you, so use at your own risk. ........ Merged revisions 427334 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d159885e50	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. Review: https://reviewboard.asterisk.org/r/2964/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	33f0251b6c	res_pjsip: Add disable_tcp_switch option. When a packet exceeds the MTU, pjproject will switch from UDP to TCP. In some circumstances (on some networks), this can cause some issues with messages not getting sent to the correct destination - and can also cause connections to get dropped due to quirks in pjproject deciding to terminate TCP connections with no messages. While fixing the routing/messaging issues is important, having a configuration option in Asterisk that tells pjproject to not switch over to TCP would be useful. That way, if some glitch is discovered on some other network/site, we can at least disable the behavior until a fix is put into place. AFS-197 #close Review: https://reviewboard.asterisk.org/r/4137/ ........ Merged revisions 427129 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427130 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ac091d4184	chan_pjsip: Add support for passing hold and unhold requests through. This change adds an option, moh_passthrough, that when enabled will pass hold and unhold requests through using a SIP re-invite. When placing on hold a re-invite with sendonly will be sent and when taking off hold a re-invite with sendrecv will be sent. This allows remote servers to handle the musiconhold instead of the local Asterisk instance being responsible. Review: https://reviewboard.asterisk.org/r/4103/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	28173ddf05	pjsip: clarify tls cert and key file usage A question arose as to whether a .pem file could be provided in place of the .crt and .key files in a PJSIP TLS configuration. I tested this and discovered that although a cert will be read from the pem file, a key will not, and thus the priv_key_file entry is still required. This update to the fine documentation clarifies the option usage. AST-1448 #close Review: https://reviewboard.asterisk.org/r/4129/ Reported by: John Bigelow ........ Merged revisions 426928 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426930 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0d0e38a0e1	res_pjsip_keepalive: Add runtime configurable keepalive module for connection-oriented transports. This change adds a module which is configurable using the keep_alive_interval setting in the global section that will send a CRLF keep alive to all active connection-oriented transports at the provided interval. This is useful because it can help keep connections open through NATs. This functionality also exists within PJSIP but can not be controlled at runtime and requires recompiling it. Review: https://reviewboard.asterisk.org/r/4084/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c6837c236f	res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprov This module allows res_pjsip to integrate with res_phoneprov. It handles the pjsip 'phoneprov' object type. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3976/ ........ Merged revisions 425007 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425008 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b1f8eba178	PJSIP: Restore functional default for callerid_privacy The pjsip config option default fixups from r424263 altered the functional default from "allowed_not_screened" to "allowed". This change restores the functional default value when none is provided. ........ Merged revisions 424426 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424427 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2b0777c017	res_pjsip: Make transport cipher option accept a comma separated list of cipher names. Improvements to the res_pjsip transport cipher option. * Made the cipher option accept a comma separated list of OpenSSL cipher names. Users of realtime will be glad if they have more than one name to list. * Added the CLI command 'pjsip list ciphers' so a user can know what OpenSSL names are available for the cipher option. * Updated the cipher option online XML documentation to specify what is expected for the value. * Updated pjsip.conf.sample to not indicate that ALL is acceptable since ALL does not imply a preference order for the ciphers and PJSIP does not simply pass the string to OpenSSL for interpretation. ASTERISK-24199 #close Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/4018/ ........ Merged revisions 424393 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424394 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	aa5458d6ab	res_pjsip: document use of rewrite_contact in sample conf Without setting rewrite_contact, an invite to an endpoint behind NAT will not reach it - unless the endpoint itself uses STUN or TURN to discover it's public URI. Thus, the use of this should be in the sample documentation. Review: https://reviewboard.asterisk.org/r/4036/ ........ Merged revisions 424337 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424338 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	adba2a8d7f	res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash. During the latest update to DTLS-SRTP support the ability to configure the hash used for fingerprints was added. This gave us two supported ones: SHA-1 and SHA-256. The default was accordingly updated to SHA-256. Unfortunately this configuration ability was not exposed within res_pjsip. This change adds a dtls_fingerprint option that controls it. #SIPit31 ........ Merged revisions 424290 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424291 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	4d2c7c23f8	PJSIP: Handle defaults properly This updates the code behind PJSIP configuration options with custom handlers to deal with the assigned default values properly where it makes sense and adjusting the default value where it doesn't. Before applying this patch, there were several cases where the default value for an option would prevent that config section from loading properly. Reported by: Thomas Thompson Review: https://reviewboard.asterisk.org/r/4019/ ........ Merged revisions 424263 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424266 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	baf99dffac	Add note about configuring list_items on a single line. ........ Merged revisions 422855 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	5ad0edacb6	Add sample configuration for resource lists. On review /r/3977, it was recommended to note in the sample configuration about the size limitation for resource lists. However, since there was no section in the sample configuration at all for resource list subscriptions, I decided to make a separate commit where I have added the necessary sample configuration as well as the size limitation warning. ........ Merged revisions 422853 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fc0fecb476	configs: Move sample config files into a subdirectory of configs This moves all samples configs from configs/ to configs/samples. This allows for additional sets of sample configuration files to be added in the future. Review: https://reviewboard.asterisk.org/r/3804/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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77 Commits (4edd24841d15ee3bb1533eb6ae7cc57a51f162bf)