Add "username" to sip show peer (bug #2163) as well as a few config cleanups
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
parent
baef4818d2
commit
deb02a000f
|
@ -5582,6 +5582,7 @@ static int sip_show_peer(int fd, int argc, char *argv[])
|
|||
ast_cli(fd, " ToHost : %s\n", peer->tohost);
|
||||
ast_cli(fd, " Addr->IP : %s Port %d\n", peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port));
|
||||
ast_cli(fd, " Defaddr->IP : %s Port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
|
||||
ast_cli(fd, " Username : %s\n", peer->username);
|
||||
ast_cli(fd, " Codecs : ");
|
||||
/* This should really be a function in frame.c */
|
||||
if (peer->capability & AST_FORMAT_G723_1)
|
||||
|
|
|
@ -22,7 +22,8 @@
|
|||
|
||||
[general]
|
||||
context=default ; Default context for incoming calls
|
||||
;recordhistory=yes ; Record SIP history by default (see sip history / sip no history)
|
||||
;recordhistory=yes ; Record SIP history by default
|
||||
; (see sip history / sip no history)
|
||||
;realm=mydomain.tld ; Realm for digest authentication
|
||||
; defaults to "asterisk"
|
||||
; Realms MUST be globally unique according to RFC 3261
|
||||
|
@ -38,12 +39,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
|
||||
;pedantic=yes ; Enable slow, pedantic checking for Pingtel
|
||||
; and multiline formatted headers for strict
|
||||
; SIP compatibility
|
||||
; SIP compatibility (defaults to "no")
|
||||
;tos=184 ; Set IP QoS to either a keyword or numeric val
|
||||
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
|
||||
;maxexpirey=3600 ; Max length of incoming registration we allow
|
||||
;defaultexpirey=120 ; Default length of incoming/outoing registration
|
||||
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
|
||||
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
|
||||
;videosupport=yes ; Turn on support for SIP video
|
||||
|
||||
;disallow=all ; First disallow all codecs
|
||||
|
@ -135,7 +136,6 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
; accountcode
|
||||
; amaflags
|
||||
; incominglimit
|
||||
; outgoinglimit
|
||||
; restrictcid
|
||||
; mailbox
|
||||
; username
|
||||
|
@ -156,31 +156,30 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
;context=from-fwd
|
||||
|
||||
;[sip_proxy-out]
|
||||
;type=peer ; we only want to call out, not be called
|
||||
;type=peer ; we only want to call out, not be called
|
||||
;secret=guessit
|
||||
;username=yourusername
|
||||
;fromuser=yourusername ; Many SIP providers require this!
|
||||
;username=yourusername ; Authentication user for outbound proxies
|
||||
;fromuser=yourusername ; Many SIP providers require this!
|
||||
;host=box.provider.com
|
||||
|
||||
;[grandstream1]
|
||||
;type=friend ; either "friend" (peer+user), "peer" or "user"
|
||||
;type=friend ; either "friend" (peer+user), "peer" or "user"
|
||||
;context=from-sip
|
||||
;username=grandstream1 ; usually matches the [section] title
|
||||
;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD
|
||||
;fromuser=grandstream1 ; overrides the callerid, e.g. required by FWD
|
||||
;callerid=John Doe <1234>
|
||||
;host=192.168.0.23 ; we have a static but private IP address
|
||||
;nat=no ; there is not NAT between phone and Asterisk
|
||||
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
|
||||
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
|
||||
;outgoinglimit=1 ; disable callwaiting signal (2nd call to phone)
|
||||
;incominglimit=1 ; permit only 1 outgoing call at a time
|
||||
;host=192.168.0.23 ; we have a static but private IP address
|
||||
;nat=no ; there is not NAT between phone and Asterisk
|
||||
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
|
||||
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
|
||||
;incominglimit=1 ; permit only 1 outgoing call at a time
|
||||
; from the phone to asterisk
|
||||
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
|
||||
;disallow=all ; need to disallow=all before we can use allow=
|
||||
;allow=ulaw ; Note: In user sections the order of codecs
|
||||
; listed with allow= does NOT matter!
|
||||
;disallow=all ; need to disallow=all before we can use allow=
|
||||
;allow=ulaw ; Note: In user sections the order of codecs
|
||||
; listed with allow= does NOT matter!
|
||||
;allow=alaw
|
||||
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
|
||||
;allow=g729 ; Pass-thru only unless g729 license obtained
|
||||
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
|
||||
;allow=g729 ; Pass-thru only unless g729 license obtained
|
||||
|
||||
|
||||
;[xlite1]
|
||||
|
@ -202,9 +201,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
|
|||
;type=friend ; Friends place calls and receive calls
|
||||
;context=from-sip ; Context for incoming calls from this user
|
||||
;secret=blah
|
||||
;language=de ; Use German prompts for this user
|
||||
;host=dynamic ; This peer register with us
|
||||
;dtmfmode=inband ; Choices are inband, rfc2833, or info
|
||||
;defaultip=192.168.0.59 ; IP used until peer registers
|
||||
;username=snom ; Username to use in INVITE until peer registers
|
||||
;mailbox=1234,2345 ; Mailboxes for message waiting indicator
|
||||
;restrictcid=yes ; To have the callerid restriced -> sent as ANI
|
||||
;disallow=all
|
||||
|
|
Loading…
Reference in New Issue