[Asterisk-Users] Why can't sip/200 call sip/202
Mark Edwards
mark.p.edwards at gmail.com
Sun Jul 24 14:14:58 MST 2005
PS you would be better seeing this debugged with "set verbose 5" in the CLI
regards,
Mark
On 7/25/05, Mark Edwards <mark.p.edwards at gmail.com> wrote:
> OK Angus
>
> just start here....
>
> mv extensions.conf extensions.conf.old
>
> and create a new extensions.conf
>
> [default]
> exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
> exten => _2XX,2,Hangup
>
>
>
> just those 3 lines
>
> do an 'extensions reload' in the CLI or just restart Asterisk
>
> and see if it works
>
> regards,
>
> Mark.
> On 7/25/05, Angus Comber <angus at iteloffice.com> wrote:
> > I think the 777 may be a bit of a Red Herring. I dialed 777 as a test. I
> > can't dial 202 from 200 if I actually dial 202!
> >
> > My extensions.conf file:
> >
> >
> > ;
> > ; Static extension configuration file, used by
> > ; the pbx_config module. This is where you configure all your
> > ; inbound and outbound calls in Asterisk.
> > ;
> > ; This configuration file is reloaded
> > ; - With the "extensions reload" command in the CLI
> > ; - With the "reload" command (that reloads everything) in the CLI
> >
> > ;
> > ; The "General" category is for certain variables.
> > ;
> > [general]
> > ;
> > ; If static is set to no, or omitted, then the pbx_config will rewrite
> > ; this file when extensions are modified. Remember that all comments
> > ; made in the file will be lost when that happens.
> > ;
> > ; XXX Not yet implemented XXX
> > ;
> > static=yes
> > ;
> > ; if static=yes and writeprotect=no, you can save dialplan by
> > ; CLI command 'save dialplan' too
> > ;
> > writeprotect=no
> >
> > ; You can include other config files, use the #include command (without the
> > ';')
> > ; Note that this is different from the "include" command that includes
> > contexts within
> > ; other contexts. The #include command works in all asterisk configuration
> > files.
> > ;#include "filename.conf"
> >
> > ; The "Globals" category contains global variables that can be referenced
> > ; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
> > variable
> > ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
> > ;
> > [globals]
> > CONSOLE=Console/dsp ; Console interface for demo
> > ;CONSOLE=Zap/1
> > ;CONSOLE=Phone/phone0
> > IAXINFO=guest ; IAXtel username/password
> > ;IAXINFO=myuser:mypass
> > TRUNK=Zap/g2 ; Trunk interface
> > ;
> > ; Note the 'g2' in the TRUNK variable above. It specifies which group
> > (defined
> > ; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use
> > in
> > ; the specified group. The four possible options are:
> > ;
> > ; g: select the lowest-numbered non-busy Zap channel (aka. ascending
> > sequential hunt group).
> > ; G: select the highest-numbered non-busy Zap channel (aka. descending
> > sequential hunt group).
> > ; r: use a round-robin search, starting at the next highest channel than
> > last time (aka. ascending rotary hunt group).
> > ; R: use a round-robin search, starting at the next lowest channel than last
> > time (aka. descending rotary hunt group).
> > ;
> > TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
> > ;TRUNK=IAX2/user:pass at provider
> >
> > ;
> > ; Any category other than "General" and "Globals" represent
> > ; extension contexts, which are collections of extensions.
> > ;
> > ; Extension names may be numbers, letters, or combinations
> > ; thereof. If an extension name is prefixed by a '_'
> > ; character, it is interpreted as a pattern rather than a
> > ; literal. In patterns, some characters have special meanings:
> > ;
> > ; X - any digit from 0-9
> > ; Z - any digit from 1-9
> > ; N - any digit from 2-9
> > ; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
> > ; . - wildcard, matches anything remaining (e.g. _9011. matches
> > ; anything starting with 9011 excluding 9011 itself)
> > ;
> > ; For example the extension _NXXXXXX would match normal 7 digit dialings,
> > ; while _1NXXNXXXXXX would represent an area code plus phone number
> > ; preceeded by a one.
> > ;
> > ; Each step of an extension is ordered by priority, which must
> > ; always start with 1 to be considered a valid extension.
> > ;
> > ; Contexts contain several lines, one for each step of each
> > ; extension, which can take one of two forms as listed below,
> > ; with the first form being preferred. One may include another
> > ; context in the current one as well, optionally with a
> > ; date and time. Included contexts are included in the order
> > ; they are listed.
> > ;
> > ;[context]
> > ;exten => someexten,priority,application(arg1,arg2,...)
> > ;exten => someexten,priority,application,arg1|arg2...
> > ;
> > ; Timing list for includes is
> > ;
> > ; <time range>|<days of week>|<days of month>|<months>
> > ;
> > ;include => daytime|9:00-17:00|mon-fri|*|*
> > ;
> > ; ignorepat can be used to instruct drivers to not cancel dialtone upon
> > ; receipt of a particular pattern. The most commonly used example is
> > ; of course '9' like this:
> > ;
> > ;ignorepat => 9
> > ;
> > ; so that dialtone remains even after dialing a 9.
> > ;
> >
> > ;
> > ; Here are the entries you need to participate in the IAXTEL
> > ; call routing system. Most IAXTEL numbers begin with 1-700, but
> > ; there are exceptions. For more information, and to sign
> > ; up, please go to www.gnophone.com or www.iaxtel.com
> > ;
> > [iaxtel700]
> > exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
> >
> > ;
> > ; The SWITCH statement permits a server to share the dialplain with
> > ; another server. Use with care: Reciprocal switch statements are not
> > ; allowed (e.g. both A -> B and B -> A), and the switched server needs
> > ; to be on-line or else dialing can be severly delayed.
> > ;
> > [iaxprovider]
> > ;switch => IAX2/user:[key]@myserver/mycontext
> >
> > [trunkint]
> > ;
> > ; International long distance through trunk
> > ;
> > exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _9011.,2,Congestion
> >
> > [trunkld]
> > ;
> > ; Long distance context accessed through trunk
> > ;
> > exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91NXXNXXXXXX,2,Congestion
> >
> > [trunklocal]
> > ;
> > ; Local seven-digit dialing accessed through trunk interface
> > ;
> > exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _9NXXXXXX,2,Congestion
> >
> > [trunktollfree]
> > ;
> > ; Long distance context accessed through trunk interface
> > ;
> > exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91800NXXXXXX,2,Congestion
> > exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91888NXXXXXX,2,Congestion
> > exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91877NXXXXXX,2,Congestion
> > exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91866NXXXXXX,2,Congestion
> >
> > [international]
> > ;
> > ; Master context for international long distance
> > ;
> > ignorepat => 9
> > include => longdistance
> > include => trunkint
> >
> > [longdistance]
> > ;
> > ; Master context for long distance
> > ;
> > ignorepat => 9
> > include => local
> > include => trunkld
> >
> > [local]
> > ;
> > ; Master context for local, toll-free, and iaxtel calls only
> > ;
> > ignorepat => 9
> > include => default
> > include => parkedcalls
> > include => trunklocal
> > include => iaxtel700
> > include => trunktollfree
> > include => iaxprovider
> > ;
> > ; You can use an alternative switch type as well, to resolve
> > ; extensions that are not known here, for example with remote
> > ; IAX switching you transparently get access to the remote
> > ; Asterisk PBX
> > ;
> > ; switch => IAX2/user:password at bigserver/local
> >
> > [macro-stdexten];
> > ;
> > ; Standard extension macro:
> > ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
> > ; ${ARG2} - Device(s) to ring
> > ;
> > exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
> > exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
> > (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
> >
> > exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
> > voicemail w/ unavail announce
> > exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start
> >
> > exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy
> > announce
> > exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start
> >
> > exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
> >
> > exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into
> > VoicemailMain
> >
> > [demo]
> > ;
> > ; We start with what to do when a call first comes in.
> > ;
> > exten => s,1,Wait,1 ; Wait a second, just for fun
> > exten => s,2,Answer ; Answer the line
> > exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
> > exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
> > exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
> > exten => s,6,BackGround(demo-instruct) ; Play some instructions
> >
> > exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
> > exten => 2,2,Goto(s,6)
> >
> > exten => 3,1,SetLanguage(fr) ; Set language to french
> > exten => 3,2,Goto(s,5) ; Start with the congratulations
> >
> > exten => 1000,1,Goto(default,s,1)
> > ;
> > ; We also create an example user, 1234, who is on the console and has
> > ; voicemail, etc.
> > ;
> > exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
> > ; (but skip if channel is not up)
> > exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
> >
> > exten => 1235,1,Voicemail(u1234) ; Right to voicemail
> >
> > exten => 1236,1,Dial(Console/dsp) ; Ring forever
> > exten => 1236,2,Voicemail(u1234) ; Unless busy
> >
> > ;
> > ; # for when they're done with the demo
> > ;
> > exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
> > exten => #,2,Hangup ; Hang them up.
> >
> > ;
> > ; A timeout and "invalid extension rule"
> > ;
> > exten => t,1,Goto(#,1) ; If they take too long, give up
> > exten => i,1,Playback(invalid) ; "That's not valid, try again"
> >
> > ;
> > ; Create an extension, 500, for dialing the
> > ; Asterisk demo.
> > ;
> > exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
> > exten => 500,2,Dial(IAX2/guest at misery.digium.com/s at default) ; Call the
> > Asterisk demo
> > exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site
> > exten => 500,4,Goto(s,6) ; Return to the start over message.
> >
> > ;
> > ; Create an extension, 600, for evaulating echo latency.
> > ;
> > exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
> > exten => 600,2,Echo ; Do the echo test
> > exten => 600,3,Playback(demo-echodone) ; Let them know it's over
> > exten => 600,4,Goto(s,6) ; Start over
> >
> > ;
> > ; Give voicemail at extension 8500
> > ;
> > exten => 8500,1,VoicemailMain
> > exten => 8500,2,Goto(s,6)
> > ;
> > ; Here's what a phone entry would look like (IXJ for example)
> > ;
> > ;exten => 1265,1,Dial(Phone/phone0,15)
> > ;exten => 1265,2,Goto(s,5)
> >
> > ;[mainmenu]
> > ;
> > ; Example "main menu" context with submenu
> > ;
> > ;exten => s,1,Answer
> > ;exten => s,2,Background(thanks) ; "Thanks for calling press 1 for sales, 2
> > for support, ..."
> > ;exten => 1,1,Goto(submenu,s,1)
> > ;exten => 2,1,Hangup
> > ;include => default
> > ;
> > ;[submenu]
> > ;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
> > ;exten => s,2,Wait,2
> > ;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales
> > department. Press 1 for steve, 2 for..."
> > ;exten => 1,1,Goto(default,steve,1)
> > ;exten => 2,1,Goto(default,mark,2)
> >
> > [default]
> > ;
> > ; By default we include the demo. In a production system, you
> > ; probably don't want to have the demo there.
> > ;
> > include => demo
> >
> > ;
> > ; Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
> > ; Note that you must have a [sipprovider] section in sip.conf whereas
> > ; the otherprovider.net example does not require such a peer definition
> > ;
> > ;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
> > ;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
> >
> > ; Real extensions would go here. Generally you want real extensions to be 4
> > or 5
> > ; digits long (although there is no such requirement) and start with a
> > single
> > ; digit that is fairly large (like 6 or 7) so that you have plenty of room
> > to
> > ; overlap extensions and menu options without conflict. You can alias them
> > with
> > ; names, too and use global variables
> >
> > ;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1 ; Channel hints for presence
> > ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
> > ;exten => 6245,1,Dial(${HINT},20,rtT) ; Use hint as listed
> > ;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit
> > ;exten => 6389,1,Dial(MGCP/aaln/1 at 192.168.0.14)
> > ;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
> >
> > ;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is
> > something like Zap/2
> > ;exten => mark,1,Goto(6275|1) ; alias mark to 6275
> > ;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
> > ;exten => wil,1,Goto(6236|1)
> > ;
> > ; Some other handy things are an extension for checking voicemail via
> > ; voicemailmain
> > ;
> > ;exten => 8500,1,VoicemailMain
> > ;exten => 8500,2,Hangup
> > ;
> > ; Or a conference room (you'll need to edit meetme.conf to enable this room)
> > ;
> > ;exten => 8600,1,Meetme(1234)
> > ;
> > ; Or playing an announcement to the called party, as soon it answers
> > ;
> > ;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
> > ;
> > ; For more information on applications, just type "show applications" at
> > your
> > ; friendly Asterisk CLI prompt.
> > ;
> > ; 'show application <command>' will show details of how you
> > ; use that particular application in this file, the dial plan.
> > ;
> >
> >
> >
> >
> > ----- Original Message -----
> > From: "dbruce" <dbruce at bananatel.ca>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > <asterisk-users at lists.digium.com>
> > Sent: Sunday, July 24, 2005 8:39 PM
> > Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
> >
> >
> > > Marc: My answer is not incorrect... it is incomplete.
> > >
> > > The OP stipulated 2 extensions 200 and 202... and provided a sip debug
> > > indicating a call from 200 to 777.
> > >
> > > I pointed out the obvious.
> > >
> > > If the OP is dialing 202 on the phone, and the phone is dialing 777, then
> > > he
> > > needs to look at the dialplan configuration of the phone. If he is dialing
> > > 777 on the phone and expecting to reach 202, then he will need to have
> > > translations in the asterisk dialplan. But, the question was "what should
> > > I
> > > be looking at?"... Using just the information provided, and the fact that
> > > he
> > > is new to asterisk... without any further information... the first thing
> > > he
> > > should be looking at is why the phone is trying to reach 777 when he wants
> > > to reach 202... Many new users do not realize the complexity of the SIP
> > > protocol, and only really look at the trace in a general manner... such
> > > as:
> > > INVITE
> > > 407 Proxy Authentication Required
> > > ACK
> > > INVITE
> > > 404 Not Found
> > > ACK
> > >
> > > The idea was to provide a clue... not to provide a complete working
> > > dialplan
> > > and phone configuration. Providing new users with "the complete package"
> > > is
> > > a dis-service to them. They will only learn from thier mistakes and
> > > experiences.. providing clues allows them to expand their experience and
> > > build their confidence... It requires them to look at the details and
> > > learn
> > > to analyse them.
> > >
> > > Regards,
> > > Derek
> > >
> > >
> > > ----- Original Message -----
> > > From: "Marc Storck" <marc.storck at msnetworks.lu>
> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > <asterisk-users at lists.digium.com>
> > > Sent: Sunday, July 24, 2005 12:53 PM
> > > Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
> > >
> > >
> > >> Derek: you reply is uncorrect. If Angus has the extension 777 in his
> > >> dialplan/extensions.conf which will dial 202. The name of the peer has
> > >> absolutely nothing to do with which number/name he would have to dial.
> > >> Without dialplan he will be unable to call any extension even 202, as
> > >> 202 is only the name of the peer.
> > >>
> > >> Angus: please paste your extensions.conf to pastebin.ca
> > >>
> > >> Regards,
> > >>
> > >> Marc
> > >>
> > >> dbruce wrote:
> > >> > It appears from the debug that extension 200 is trying to call 777, not
> > >> > 202. Your Asterisk server can't find an extension 777 and returns "404
> > >> > not found". That will explain why you can't call extension 777 from
> > >> > extension 200. If you want to call extension 202, you will need to dial
> > >> > 202 on extension 200, not 777.
> > >> >
> > >> > Regards,
> > >> > Derek
> > >> >
> > >> >
> > >> > ----- Original Message -----
> > >> > *From:* Angus Comber <mailto:angus at iteloffice.com>
> > >> > *To:* asterisk-users at lists.digium.com
> > >> > <mailto:asterisk-users at lists.digium.com>
> > >> > *Sent:* Sunday, July 24, 2005 11:51 AM
> > >> > *Subject:* [Asterisk-Users] Why can't sip/200 call sip/202
> > >> >
> > >> > I have 2 sip accounts setup - 200 and 202. If I do sip show peers
> > >> > I
> > >> > get:
> > >> >
> > >> > sip show peers
> > >> > Name/username Host Dyn Nat ACL Mask
> > >> > Port Status
> > >> > 202/202 192.168.0.6 D 255.255.255.255
> > >> > 5060 Unmonitored
> > >> > 201/201 (Unspecified) D 255.255.255.255
> > >> > 5060 Unmonitored
> > >> > 200/200 192.168.0.3 D 255.255.255.255
> > >> > 5060 Unmonitored
> > >> >
> > >> > 200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream BT100
> > >> > IP phone.
> > >> >
> > >> > relevant bit of sip.conf:
> > >> >
> > >> > [200]
> > >> > username=200
> > >> > type=friend
> > >> > secret=1234
> > >> > port=5060
> > >> > nat=never
> > >> > dtmfmode=rfc2833
> > >> > context=default
> > >> > callerid="Angus Comber" <200>
> > >> > host=dynamic
> > >> > disallow=all
> > >> > allow=ulaw
> > >> > allow=alaw
> > >> > allow=g723.1
> > >> > allow=g729
> > >> >
> > >> > [202]
> > >> > username=202
> > >> > type=friend
> > >> > secret=1234
> > >> > port=5060
> > >> > nat=never
> > >> > dtmfmode=rfc2833
> > >> > context=default
> > >> > callerid="Sam Comber" <202>
> > >> > host=dynamic
> > >> > disallow=all
> > >> > allow=ulaw
> > >> > allow=alaw
> > >> > allow=g723.1
> > >> > allow=g729
> > >> >
> > >> >
> > >> > But whenever I try to dial between phones I get this:
> > >> >
> > >> >
> > >> > Sip read:
> > >> >
> > >> > 0 headers, 0 lines
> > >> >
> > >> >
> > >> > Sip read:
> > >> > INVITE sip:777 at 192.168.0.13;user=phone SIP/2.0
> > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
> > >> > From: "Angus Comber"
> > >> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > >> > To: <sip:777 at 192.168.0.13;user=phone>
> > >> > Contact: <sip:200 at 192.168.0.3;user=phone>
> > >> > Supported: replaces, timer
> > >> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > >> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > >> > CSeq: 45925 INVITE
> > >> > User-Agent: Grandstream GXP2000 1.0.1.9
> > >> > Max-Forwards: 70
> > >> > Allow:
> > >> >
> > > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> > >> > Content-Type: application/sdp
> > >> > Content-Length: 258
> > >> >
> > >> > v=0
> > >> > o=200 8000 8000 IN IP4 192.168.0.3
> > >> > s=SIP Call
> > >> > c=IN IP4 192.168.0.3
> > >> > t=0 0
> > >> > m=audio 5004 RTP/AVP 18 0 8 101
> > >> > a=sendrecv
> > >> > a=rtpmap:18 G729/8000
> > >> > a=rtpmap:0 PCMU/8000
> > >> > a=rtpmap:8 PCMA/8000
> > >> > a=ptime:20
> > >> > a=rtpmap:101 telephone-event/8000
> > >> > a=fmtp:101 0-11
> > >> >
> > >> > 13 headers, 13 lines
> > >> > Using latest request as basis request
> > >> > Sending to 192.168.0.3 : 5060 (non-NAT)
> > >> > Reliably Transmitting (no NAT):
> > >> > SIP/2.0 407 Proxy Authentication Required
> > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
> > >> > From: "Angus Comber"
> > >> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > >> > To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
> > >> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > >> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > >> > CSeq: 45925 INVITE
> > >> > User-Agent: Asterisk PBX
> > >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > >> > Contact: <sip:777 at 192.168.0.13>
> > >> > Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366"
> > >> > Content-Length: 0
> > >> >
> > >> >
> > >> > to 192.168.0.3:5060
> > >> > Scheduling destruction of call '11e4ca07b25c9335 at 192.168.0.3'
> > >> > <mailto:'11e4ca07b25c9335 at 192.168.0.3'> in 15000 ms
> > >> > Found user '200'
> > >> >
> > >> >
> > >> > Sip read:
> > >> > ACK sip:777 at 192.168.0.13;user=phone SIP/2.0
> > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
> > >> > From: "Angus Comber"
> > >> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > >> > To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
> > >> > Contact: <sip:200 at 192.168.0.3;user=phone>
> > >> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > >> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > >> > CSeq: 45925 ACK
> > >> > User-Agent: Grandstream GXP2000 1.0.1.9
> > >> > Max-Forwards: 70
> > >> > Allow:
> > >> >
> > > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> > >> > Content-Length: 0
> > >> >
> > >> >
> > >> > 11 headers, 0 lines
> > >> >
> > >> >
> > >> > Sip read:
> > >> > INVITE sip:777 at 192.168.0.13;user=phone SIP/2.0
> > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
> > >> > From: "Angus Comber"
> > >> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > >> > To: <sip:777 at 192.168.0.13;user=phone>
> > >> > Contact: <sip:200 at 192.168.0.3;user=phone>
> > >> > Supported: replaces, timer
> > >> > Proxy-Authorization: Digest username="200", realm="asterisk",
> > >> > algorithm=MD5, uri="sip:777 at 192.168.0.13;user=phone",
> > >> > nonce="0c555366", response="ee6088fb4e50da5fe412913ae40dd45c"
> > >> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > >> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > >> > CSeq: 45926 INVITE
> > >> > User-Agent: Grandstream GXP2000 1.0.1.9
> > >> > Max-Forwards: 70
> > >> > Allow:
> > >> >
> > > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> > >> > Content-Type: application/sdp
> > >> > Content-Length: 258
> > >> >
> > >> > v=0
> > >> > o=200 8000 8001 IN IP4 192.168.0.3
> > >> > s=SIP Call
> > >> > c=IN IP4 192.168.0.3
> > >> > t=0 0
> > >> > m=audio 5004 RTP/AVP 18 0 8 101
> > >> > a=sendrecv
> > >> > a=rtpmap:18 G729/8000
> > >> > a=rtpmap:0 PCMU/8000
> > >> > a=rtpmap:8 PCMA/8000
> > >> > a=ptime:20
> > >> > a=rtpmap:101 telephone-event/8000
> > >> > a=fmtp:101 0-11
> > >> >
> > >> > 14 headers, 13 lines
> > >> > Using latest request as basis request
> > >> > Sending to 192.168.0.3 : 5060 (non-NAT)
> > >> > Found user '200'
> > >> > Found RTP audio format 18
> > >> > Found RTP audio format 0
> > >> > Found RTP audio format 8
> > >> > Found RTP audio format 101
> > >> > Peer audio RTP is at port 192.168.0.3:5004
> > >> > Found description format G729
> > >> > Found description format PCMU
> > >> > Found description format PCMA
> > >> > Found description format telephone-event
> > >> > Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10c
> > >> > (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c
> > > (ulaw|alaw|g729)
> > >> > Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723),
> > >> > combined
> > >> > - 0x1 (g723)
> > >> > Looking for 777 in default
> > >> > Reliably Transmitting (no NAT):
> > >> > SIP/2.0 404 Not Found
> > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
> > >> > From: "Angus Comber"
> > >> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > >> > To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
> > >> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > >> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > >> > CSeq: 45926 INVITE
> > >> > User-Agent: Asterisk PBX
> > >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > >> > Contact: <sip:777 at 192.168.0.13>
> > >> > Content-Length: 0
> > >> >
> > >> >
> > >> > to 192.168.0.3:5060
> > >> >
> > >> >
> > >> > Sip read:
> > >> > ACK sip:777 at 192.168.0.13;user=phone SIP/2.0
> > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
> > >> > From: "Angus Comber"
> > >> > <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
> > >> > To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
> > >> > Contact: <sip:200 at 192.168.0.3;user=phone>
> > >> > Proxy-Authorization: Digest username="200", realm="asterisk",
> > >> > algorithm=MD5, uri="sip:777 at 192.168.0.13;user=phone",
> > >> > nonce="0c555366", response="7fcb1024a81b3ea3bcc56baeca4bac3e"
> > >> > Call-ID: 11e4ca07b25c9335 at 192.168.0.3
> > >> > <mailto:11e4ca07b25c9335 at 192.168.0.3>
> > >> > CSeq: 45926 ACK
> > >> > User-Agent: Grandstream GXP2000 1.0.1.9
> > >> > Max-Forwards: 70
> > >> > Allow:
> > >> >
> > > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> > >> > Content-Length: 0
> > >> >
> > >> >
> > >> > 12 headers, 0 lines
> > >> > Destroying call '11e4ca07b25c9335 at 192.168.0.3'
> > >> > <mailto:'11e4ca07b25c9335 at 192.168.0.3'>
> > >> >
> > >> >
> > >> > How can I troubleshoot? What should I be looking at?
> > >> >
> > >> > Angus
> > >> >
> > >> >
> > >>
> > > ------------------------------------------------------------------------
> > >> >
> > >> > _______________________________________________
> > >> > Asterisk-Users mailing list
> > >> > Asterisk-Users at lists.digium.com
> > >> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >> > To UNSUBSCRIBE or update options visit:
> > >> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >> >
> > >> >
> > >> > ------------------------------------------------------------------------
> > >> >
> > >> > _______________________________________________
> > >> > Asterisk-Users mailing list
> > >> > Asterisk-Users at lists.digium.com
> > >> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >> > To UNSUBSCRIBE or update options visit:
> > >> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>
> > >> --
> > >> CTO Marc Storck
> > >> MS Networks SA mstorck at msnetworks.lu
> > >> IT Service Provider http://www.msnetworks.lu
> > >> 15, route d'Esch Phone: +352 2727 3030
> > >> L-4450 Belvaux Fax: +352 2727 3060
> > >>
> > >> --------------- MS Networks powered service ---------------
> > >> http://www.LuxAdmin.com Hosting and housing solutions
> > >> -----------------------------------------------------------
> > >>
> > >> _______________________________________________
> > >> Asterisk-Users mailing list
> > >> Asterisk-Users at lists.digium.com
> > >> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >> To UNSUBSCRIBE or update options visit:
> > >> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> regards,
>
> Mark P. Edwards
> FWD: 667917
>
--
regards,
Mark P. Edwards
FWD: 667917
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