SIPp는 Asterisk에 "hello world" 메시지를 보냅니다.

Asterisk에 대해 진단을 실행하는 방법은 무엇입니까? 별표는 동일한 네트워크에서 실행 중 tleilax입니다 (doge내 네트워크 토폴로지가 최적이 아닙니다.).

구체적으로 다음과 같은 작업을 수행하고 싶습니다.

sipp [email protected]

하지만 어떤 플래그를 보내야 할지 잘 모르겠습니다. "hello world"를 어떻게 보내나요 [email protected]?

(이 모든 내용은 내 LAN에 있으며 인터넷을 통해 액세스할 수 없습니다.)

한 모금 구성 파일:

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:~ # 

십삭지역적 성공:

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:~$ 

십삭실패했습니다. 홉이 너무 많습니다.

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:~$ 

Ekiga 또는 기타 GUI 소프트폰을 Asterisk에 연결하려고 할 때입니다..