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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:
Mark Spencer 2004-07-28 21:07:38 +00:00
parent baef4818d2
commit deb02a000f
2 changed files with 23 additions and 21 deletions
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1
channels/chan_sip.c
View File

@ -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)

43
configs/sip.conf.sample
View File

@ -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