[Asterisk-Users] Why can't sip/200 call sip/202
Angus Comber
angus at iteloffice.com
Sun Jul 24 14:16:19 MST 2005
Would this do it:
exten => _2XX,1,Dial(${ARG1},30)
Then I would fallback to voicemail (or something else) after the 30 seconds?
Angus
----- 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 10:06 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
> Ok your extensions.conf doesn't mention anything about an extension/number
> equal to 202 or 200. You must know that the name of a SIP and IAX2 peer is
> only an "address", you will have to assign a number via extensions.conf to
> this address.
>
> Have a look at www.voip-info.org and of course google.com to get to know
> extensions.conf.
>
> Regards,
>
> Marc
>
> Angus Comber 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
>
> --
> 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
>
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