How do I run diagnostics against Asterisk?  Asterisk is running on tleilax; and doge is on the same network (My network topology isn't optimal).Specifically, I want to do something like:sipp 345@tleilax.bounceme.netexcept I'm not sure what flags to send.  How do I send hello world to 345@tleilax.bounceme.net?  (Note that this is all on my LAN, and not accessible through the internet.)sip.conf:tleilax:~ # tleilax:~ # cat /etc/asterisk/sip.conf[general]context=trunkinbound            ; Default context for incoming callsallowguest=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 authenticationbindport=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 thedisallow=all                    ; First disallow all codecsallow=ulaw                      ; Allow codecs in order of preferenceallow=gsmmohinterpret=defaultmohsuggest=defaultlanguage=en                     ; Default language setting for all users/peersrelaxdtmf=yes                   ; Relax dtmf handlingtrustrpid = no                  ; If Remote-Party-ID should be trustedsendrpid = yes                  ; If Remote-Party-ID should be sentprogressinband=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 containsdtmfmode = 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=sipregistrationsrtptimeout=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 requestsnotifyringing = 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:password@mysipprovider.com;registertimeout=20             ; retry registration calls every 20 seconds (default);registerattempts=10            ; Number of registration attempts before we give upexternip = 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 iflocalnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networkslocalnet=10.0.0.0/255.0.0.0     ; Also RFC1918localnet=172.16.0.0/12          ; Another RFC1918 with CIDR notationlocalnet=169.254.0.0/255.255.0.0 ;Zero conf local networknat=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 tojbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of ajbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIPjbmaxsize = 100             ; Max length of the jitterbuffer in milliseconds.jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer isjbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIPjblog = no                  ; Enables jitterbuffer frame logging. Defaults to no.qualify=yes             ; By default, qualify all peers at 2000mslimitonpeer = 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:test@10.10.10.16:5060;; 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=1000tleilax:~ # 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=101secret=passwordaccountcode=101callerid= <101>mailbox=101context=defaulttype=friendhost=dynamic[gs102]username=gs102secret=passwordaccountcode=gs102callerid=Test Admin Phone <>mailbox=102context=defaulttype=friendhost=dynamic; END OF FILE    Last Forced System Reload: 2015-02-20 16:49:28tleilax:~ # tleilax:~ # sipsak local success:thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:345@tleilax -m hiNo SRV record: _sip._tcp.tleilaxNo SRV record: _sip._udp.tleilaxusing A record: tleilaxMax-Forwards set to 0message received:SIP/2.0 200 OKVia: SIP/2.0/UDP 127.0.1.1:59012;branch=z9hG4bK.61911e9a;alias;received=192.168.1.3;rport=59012From: sip:sipsak@127.0.1.1:59012;tag=1c498905To: sip:345@tleilax;tag=as0e771d06Call-ID: 474581253@127.0.1.1CSeq: 1 OPTIONSServer: Asterisk PBX 1.8.29.0-viciAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContact: <sip:192.168.1.2:5060>Accept: application/sdpContent-Length: 0** reply received after 0.830 ms **   SIP/2.0 200 OK   final receivedthufir@doge:~$ sipsak failure; too many hops:thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:thufir@ekiga.net -m hiNo SRV record: _sip._tcp.ekiga.netNo SRV record: _sip._udp.ekiga.netusing A record: ekiga.netMax-Forwards set to 0message received:SIP/2.0 483 Too Many HopsVia: SIP/2.0/UDP 192.168.1.3:55929;branch=z9hG4bK.3f8863cd;rport=55929;alias;received=96.48.128.162From: sip:sipsak@127.0.1.1:55929;tag=3feca6b3To: sip:thufir@ekiga.net;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.2949Call-ID: 1072473779@127.0.1.1CSeq: 1 OPTIONSServer: Kamailio (1.5.3-notls (i386/linux))Content-Length: 0** reply received after 155.411 ms **   SIP/2.0 483 Too Many Hops   final receivedthufir@doge:~$ This is in the context of trying to connect Ekiga, or other GUI softphone, to Asterisk.
SIPp hello world message to Asterisk
networking;command line;asterisk;voip;sip
You should use domain from your realm,not ekigaYou can troubelshoot by usingasterisk -rsip set debug on