asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	15aa3c0a23	chan_sip: Add rtcp-mux support ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639	8 years ago
Richard Mudgett	4aaa27e532	Sample configs: Eliminate false multiline comment block starts. Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6	9 years ago
Alexander Traud	1a9555f036	sip.conf: tlsclientmethod is using sslv23 as default. When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL SSLv23_method. This was documented incorrectly in the file sip.conf.sample. SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that function should have been called 'secure_method' or 'automatic_method' back in the 90s. Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if you face a server which has problems like not falling back to TLSv1.0 automatically. ASTERISK-24425 Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3	9 years ago
Timo Teräs	538c6415c6	chan_sip: Support auth username for callbackextension feature ASTERISK-20527 #close Change-Id: I659cf7f00836a09d09d146ad226a40477d731239	9 years ago
Leif Madsen	6ede210c98	Remove reference to non-existent sip.conf option Option was removed in commit `7f883ef495` ASTERISK-25927 #close Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8	9 years ago
Walter Doekes	c00082329e	chan_sip: Optionally supply fromuser/fromdomain in SIP dial string. Previously you could add [!dnid] to the SIP dial string to alter the To: header. This change allows you to alter the From header as well. SIP dial string extra options now look like this: [![touser[@todomain]][![fromuser][@fromdomain]]] INCOMPATIBLE CHANGE: If you were using an exclamation mark in your To: header, that is no longer possible. ASTERISK-25803 #close Change-Id: I2457e9ba7a89eb1da22084bab5a4d4328e189db7	9 years ago
Ward van Wanrooij	d4b10cfb3e	chan_sip: option 'notifyringing' change and doc fix In the sample sip.conf this is written with regard to notifyringing: ;notifyringing = no ; Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes) However, this setting changes whether or not any RINGING indications are sent to subscriptions. There is no separate configurable setting that allows to control whether INUSE subscriptions also get sent RINGING. This is however a useful option, to see (using BLF) if somebody else is able to handle an incoming call or if everybody is busy. This patch corrects the documentation for notifyringing (so the documentation matches the functionality) and make notifyringing a tri-state option, by adding the value 'notinuse' (in addition to 'yes' and 'no'). When notifyringing = notinuse, only subscriptions that are not INUSE are sent the RINGING signal. The default setting for notifyringing remains set to yes, so the default behaviour is not affected. ASTERISK-25558 Change-Id: I88f7036ee084bb3f43b74f15612695c6708f74aa	9 years ago
Corey Farrell	40574a2ea3	chan_sip: Allow websockets to be disabled. This patch adds a new setting "websockets_enabled" to sip.conf. Setting this to false allows chan_sip to be used without causing conflicts with res_pjsip_transport_websocket. ASTERISK-24106 #close Reported by: Andrew Nagy Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7	10 years ago
Rusty Newton	d02196448b	Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c * In sip.conf.sample fix sentence where we said that WS or WSS are supported transports for use in an outbound register definition. They are not supported in that case. * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used to enable CDR on a channel. ASTERISK-24867 #close Reported by: Rusty Newton ASTERISK-24853 #close Reported by: PSDK Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca	10 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	10 years ago
Kevin Harwell	66f3fd0028	chan_sip: make progressinband default to no After the "progressinband" value setting of "never" was updated to never send a 183 this separated its use from the "no" value. Since "never" was the default, but most users probably expect "no" this patch updates the default for the "progressinband" setting to "no." ASTERISK-24835 #close Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4606/ ........ Merged revisions 434654 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Olle Johansson	d663e045f5	sip.conf.sample - note that media_address does not change listen address, just the SDP git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d0523b4b3c	chan_sip: Add support for setting DTLS configuration in the general section. Configuration of DTLS in the general section will be applied to any users or peers. If configuration exists at their level it overrides the general section values. ASTERISK-24128 #close Reported by: Michael K. patches: dtls_default_settings.patch submitted by Michael K. (license 6621) Review: https://reviewboard.asterisk.org/r/3867/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	a62fedf0cb	chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI. Document it in sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged revisions 423066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423067 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423068 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423069 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Paul Belanger	ef28cc0d43	chan_sip.c: Add 'rtpbindaddr' setting Users now have the ability to bind the rtpengine instance to a specific IP address. For example, you want chan_sip (call control) on eth0 but rtp (media) on eth1. ASTERISK-24280 #close Reported by: Paul Belanger Tested by: Paul Belanger Review: https://reviewboard.asterisk.org/r/3952/ Patches: rtpengine.diff uploaded by Paul Belanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422241 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

16 Commits (c1ab8ca74cec45730107cca3ed47fc61460365e4)