[Asterisk-Users] Why can't sip/200 call sip/202
Marc Storck
marc.storck at msnetworks.lu
Sun Jul 24 14:06:54 MST 2005
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
-----------------------------------------------------------
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