[Asterisk-Users] Why can't sip/200 call sip/202
Marc Storck
marc.storck at msnetworks.lu
Sun Jul 24 14:42:43 MST 2005
No please use ${EXTEN}, ${ARG1} is for macros.
And of course you will use the protocol in front of ${EXTEN}
So for SIP use:
exten => _2XX,1,Dial(SIP/${EXTEN},30)
and for IAX2 use:
exten => _2XX,1,Dial(IAX2/${EXTEN},30)
Regards,
Marc
Angus Comber wrote:
> Would this do it:
>
> exten => _2XX,1,Dial(${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
>>
>
>
> _______________________________________________
> 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
-----------------------------------------------------------
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