diff options
author | bbryant <bbryant@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-09-09 18:51:52 +0000 |
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committer | bbryant <bbryant@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-09-09 18:51:52 +0000 |
commit | 722eb3c4c3cfa1c0cee915c949c5f95199ee24dd (patch) | |
tree | 25683963c5e51bdedd6211cd0ea92a85639505c3 /configs/sip.conf.sample | |
parent | 815b5b09da5e555add7bba3d8fca588e7611248a (diff) |
Merged revisions 285710 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
........
r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/sip.conf.sample')
-rw-r--r-- | configs/sip.conf.sample | 55 |
1 files changed, 37 insertions, 18 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 01fd29b00..320895669 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -144,6 +144,8 @@ allowoverlap=no ; Disable overlap dialing support. (Default is y ; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=:: ; (You can choose independently for UDP, TCP, and TLS, by specifying different values for ; "udpbindaddr", "tcpbindaddr", and "tlsbindaddr".) +; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat. +; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.) ; ; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061 ; for TLS). @@ -213,7 +215,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;pedantic=yes ; Enable checking of tags in headers, ; international character conversions in URIs ; and multiline formatted headers for strict - ; SIP compatibility (defaults to "no") + ; SIP compatibility (defaults to "yes") ; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters. ;tos_sip=cs3 ; Sets TOS for SIP packets. @@ -253,6 +255,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Message-Account in the MWI notify message ; defaults to "asterisk" +; Codec negotiation +; +; When Asterisk is receiving a call, the codec will initially be set to the +; first codec in the allowed codecs defined for the user receiving the call +; that the caller also indicates that it supports. But, after the caller +; starts sending RTP, Asterisk will switch to using whatever codec the caller +; is sending. +; +; When Asterisk is placing a call, the codec used will be the first codec in +; the allowed codecs that the callee indicates that it supports. Asterisk will +; *not* switch to whatever codec the callee is sending. +; ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec ; rather than advertising all joint codec capabilities. This ; limits the other side's codec choice to exactly what we prefer. @@ -354,6 +368,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; instead of letting the requester know whether there was ; a matching user or peer for their request. This reduces ; the ability of an attacker to scan for valid SIP usernames. + ; This option is set to "yes" by default. + +;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like + ; INVITE requests are. By default this option is disabled. ;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing ; order instead of RFC3551 packing order (this is required @@ -364,6 +382,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices ;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers ;outboundproxy=tls://proxy.provider.domain ; same as '=proxy.provider.domain' except we try to connect with tls +;outboundproxy=192.0.2.1 ; IPv4 address literal (default port is 5060) +;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060) +;outboundproxy=192.168.0.2.1:5062 ; IPv4 address literal with explicit port +;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port ; ; (could also be tcp,udp) - defining transports on the proxy line only ; ; applies for the global proxy, otherwise use the transport= option ;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches @@ -720,31 +742,18 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; externhost=foo.dyndns.net ; refreshed periodically ; externrefresh=180 ; change the refresh interval ; -; c. "stunaddr = stun.server[:port]" queries the STUN server specified -; as an argument to obtain the external address/port. -; Queries are also sent periodically every "externrefresh" seconds -; (as a side effect, sending the query also acts as a keepalive for -; the state entry on the nat box): -; -; stunaddr = foo.stun.com:3478 -; externrefresh = 15 -; -; NOTE: STUN is only implemented for IPv4. -; ; Note that at the moment all these mechanism work only for the SIP socket. -; The IP address discovered with externaddr/externhost/STUN is reused for +; The IP address discovered with externaddr/externhost is reused for ; media sessions as well, but the port numbers are not remapped so you ; may still experience problems. ; ; NOTE 1: in some cases, NAT boxes will use different port numbers in ; the internal<->external mapping. In these cases, the "externaddr" and -; "externhost" might not help you configure addresses properly, and you -; really need to use STUN. +; "externhost" might not help you configure addresses properly. ; ; NOTE 2: when using "externaddr" or "externhost", the address part is -; also used as the external address for media sessions. Even if you -; use "stunaddr", STUN queries will be sent only from the SIP port, -; not from media sockets. Thus, the port information in the SDP may be wrong! +; also used as the external address for media sessions. Thus, the port +; information in the SDP may be wrong! ; ; In addition to the above, Asterisk has an additional "nat" parameter to ; address NAT-related issues in incoming SIP or media sessions. @@ -776,6 +785,16 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; can not be set per-user or per-peer. ; ; media_address = 172.16.42.1 +; +; Through the use of the res_stun_monitor module, Asterisk has the ability to detect when the +; perceived external network address has changed. When the stun_monitor is installed and +; configured, chan_sip will renew all outbound registrations when the monitor detects any sort +; of network change has occurred. By default this option is enabled, but only takes effect once +; res_stun_monitor is configured. If res_stun_monitor is enabled and you wish to not +; generate all outbound registrations on a network change, use the option below to disable +; this feature. +; +; subscribe_network_change_event = yes ; on by default ;----------------------------------- MEDIA HANDLING -------------------------------- ; By default, Asterisk tries to re-invite media streams to an optimal path. If there's |