[Asterisk-Users] Why can't sip/200 call sip/202
Angus Comber
angus at iteloffice.com
Sun Jul 24 14:30:38 MST 2005
You observed correctly. Yes I just copied the sample file, hoping it would
work.
I didn't realise I had to do anything special with the dialplan just for
dialing internal extensions.
Can I use something fairly generic like this (assuming all my extensions are
three digit starting with 2xx):
exten => _2XX,1,Dial(${ARG1})
As a VERY basic first attempt.
By the way can I use (${ARG1}) - is it valid? Or some other variable name
for number dialed?
Is there an Asterisk document on the dialplan. Eg all the variables such as
Dial, Voicemail, etc? Or do we need to look in a certain .h file?
Angus
----- 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 10:10 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
> The extensions.conf file you provided looks suspiciously like the asterisk
> configs/extensions.conf.sample file.
>
> Did you create a dialplan for your specific configuration or did you just
> copy the sample file?
>
>
>
> ----- Original Message -----
> From: "Angus Comber" <angus at iteloffice.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Sunday, July 24, 2005 2:50 PM
> Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
>
>
>> 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
>> >
>>
>>
>> _______________________________________________
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>> 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
>
> _______________________________________________
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