Asteriskに対して診断を実行する方法は？アスタリスクは同じネットワークで実行されていますtleilax（doge私のネットワークトポロジは最適ではありません。）。
具体的には、次のようなことをしたいと思います。
sipp [email protected]
しかし、どのフラグを送るべきかわかりません。 「hello world」を送るには[email protected]？
（これはすべて私のLANにあり、インターネットを介してアクセスすることはできません。)
1つの募金設定ファイル:
tleilax:~ #
tleilax:~ # cat /etc/asterisk/sip.conf
[general]
context=trunkinbound ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld ; Set default domain for this host
;pedantic=yes ; Enable checking of tags in headers,
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes ; send compact sip headers.
videosupport=no ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes ; generate manager events when sip ua
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes ; Turn on SIP debugging by default, from
;recordhistory=yes ; Record SIP history by default
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes ; Default false
;register => 1234:[email protected]
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
externip = 96.48.128.162 ; Address that we're going to put in outbound SIP
;externhost=test.test.com ; Alternatively you can specify a domain
;externrefresh=10 ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=yes ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes ; Save systemname in realtime database at registration
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes ; Enabling this setting has two functions:
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4 ; Add IP address as local domain
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
;autodomain=yes ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers
#include sip-vicidial.conf
; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:[email protected]:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000
tleilax:~ #
sip-vicidial.conf:
tleilax:~ #
tleilax:~ # cat /etc/asterisk/sip-vicidial.conf
; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
[101]
username=101
secret=password
accountcode=101
callerid="" <101>
mailbox=101
context=default
type=friend
host=dynamic
[gs102]
username=gs102
secret=password
accountcode=gs102
callerid="Test Admin Phone" <>
mailbox=102
context=default
type=friend
host=dynamic
; END OF FILE Last Forced System Reload: 2015-02-20 16:49:28
tleilax:~ #
tleilax:~ #
10地域の成功：
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:345@tleilax -m "hi"
No SRV record: _sip._tcp.tleilax
No SRV record: _sip._udp.tleilax
using A record: tleilax
Max-Forwards set to 0
message received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.1.1:59012;branch=z9hG4bK.61911e9a;alias;received=192.168.1.3;rport=59012
From: sip:[email protected]:59012;tag=1c498905
To: sip:345@tleilax;tag=as0e771d06
Call-ID: [email protected]
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.29.0-vici
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.1.2:5060>
Accept: application/sdp
Content-Length: 0
** reply received after 0.830 ms **
SIP/2.0 200 OK
final received
thufir@doge:~$
10失敗しました。ホップが多すぎます。
thufir@doge:~$
thufir@doge:~$ sudo sipsak -vv -s sip:[email protected] -m "hi"
No SRV record: _sip._tcp.ekiga.net
No SRV record: _sip._udp.ekiga.net
using A record: ekiga.net
Max-Forwards set to 0
message received:
SIP/2.0 483 Too Many Hops
Via: SIP/2.0/UDP 192.168.1.3:55929;branch=z9hG4bK.3f8863cd;rport=55929;alias;received=96.48.128.162
From: sip:[email protected]:55929;tag=3feca6b3
To: sip:[email protected];tag=c64e1f832a41ec1c1f4e5673ac5b80f6.2949
Call-ID: [email protected]
CSeq: 1 OPTIONS
Server: Kamailio (1.5.3-notls (i386/linux))
Content-Length: 0
** reply received after 155.411 ms **
SIP/2.0 483 Too Many Hops
final received
thufir@doge:~$
答え1
ekiga以外のドメインのドメイン名を使用する必要があります。
次の方法を使用して問題を解決できます。
asterisk -r
sip set debug on