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path: root/configs/samples/sip.conf.sample
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2017-11-06dtls: Add support for ephemeral DTLS certificates.Sean Bright
This mimics the behavior of Chrome and Firefox and creates an ephemeral X.509 certificate for each DTLS session. Currently, the only supported key type is ECDSA because of its faster generation time, but other key types can be added in the future as necessary. ASTERISK-27395 Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4
2017-03-17chan_sip: Add rtcp-mux supportSean Bright
ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639
2016-09-02Sample configs: Eliminate false multiline comment block starts.Richard Mudgett
Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6
2016-08-19sip.conf: tlsclientmethod is using sslv23 as default.Alexander Traud
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
2016-06-03chan_sip: Support auth username for callbackextension featureTimo Teräs
ASTERISK-20527 #close Change-Id: I659cf7f00836a09d09d146ad226a40477d731239
2016-04-22Remove reference to non-existent sip.conf optionLeif Madsen
Option was removed in commit 7f883ef495b57ae9182e47213d01d5e8009dbf3f ASTERISK-25927 #close Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8
2016-02-19chan_sip: Optionally supply fromuser/fromdomain in SIP dial string.Walter Doekes
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
2015-12-26chan_sip: option 'notifyringing' change and doc fixWard van Wanrooij
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
2015-11-03chan_sip: Allow websockets to be disabled.Corey Farrell
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
2015-07-20Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.cRusty Newton
* 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
2015-05-15tcptls: Enable multiple TLS certificate chains (RSA+ECC+DSA) for server socket.Alexander Traud
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
2015-04-10chan_sip: make progressinband default to noKevin Harwell
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
2014-11-21sip.conf.sample - note that media_address does not change listen address, ↵Olle Johansson
just the SDP git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-11-15chan_sip: Add support for setting DTLS configuration in the general section.Joshua Colp
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
2014-09-14chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI.Walter Doekes
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
2014-08-28chan_sip.c: Add 'rtpbindaddr' settingPaul Belanger
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
2014-07-17configs: Move sample config files into a subdirectory of configsMatthew Jordan
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
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