[Asterisk-Users] Why can't sip/200 call sip/202

Angus Comber angus at iteloffice.com
Sun Jul 24 15:59:37 MST 2005


That was another problem - now fixed.

Thanks for all your help on extensions.conf

Angus

----- Original Message ----- 
From: "Angus Comber" <angus at iteloffice.com>
To: "Mark Edwards" <mark.p.edwards at gmail.com>; "Asterisk Users Mailing 
List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Sent: Sunday, July 24, 2005 11:30 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202


> Sorry to be a pain... but
>
> I restarted my * and now when I launch * get this:
>
>  == Parsing '/etc/asterisk/zapata.conf': Found
> Jul 24 18:52:45 WARNING[6817]: chan_zap.c:932 zt_open: Unable to specify 
> channel 1: No such device or address
> Jul 24 18:52:45 ERROR[6817]: chan_zap.c:6473 mkintf: Unable to open 
> channel 1: No such device or address
> here = 0, tmp->channel = 1, channel = 1
> Jul 24 18:52:45 ERROR[6817]: chan_zap.c:10303 setup_zap: Unable to 
> register channel '1-2'
> Jul 24 18:52:45 WARNING[6817]: loader.c:345 ast_load_resource: 
> chan_zap.so: load_module failed, returning -1
>  == Unregistered channel type 'Tor'
>  == Unregistered channel type 'Zap'
> Jul 24 18:52:45 WARNING[6817]: loader.c:440 load_modules: Loading module 
> chan_zap.so failed!
> linux:~ # Ouch ... error while writing audio data: : Broken pipe
>
> I have a Junghanns quadBRI card installed.  I have modprobe qozap - so it 
> is loaded and seems to be working OK.  I assume there was something in 
> extensions.conf which was somehow required.  something to do with 
> [channels] ?
>
> The error in chan_zap.c seems to be saying that channel 1 cannot be 
> opened.
>
> Angus
>
>
>
>
> ----- Original Message ----- 
> From: "Mark Edwards" <mark.p.edwards at gmail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users at lists.digium.com>
> Sent: Sunday, July 24, 2005 10:13 PM
> Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
>
>
> OK Angus
>
> just start here....
>
> mv extensions.conf extensions.conf.old
>
> and create a new extensions.conf
>
> [default]
> exten => _2XX,1,Dial(SIP/${EXTEN},20,Ttm)
> exten => _2XX,2,Hangup
>
>
>
> just those 3 lines
>
> do an 'extensions reload' in the CLI or just restart Asterisk
>
> and see if it works
>
> regards,
>
> Mark.
> On 7/25/05, Angus Comber <angus at iteloffice.com> wrote:
>> I think the 777 may be a bit of a Red Herring.  I dialed 777 as a test. 
>> I
>> can't dial 202 from 200 if I actually dial 202!
>>
>> My extensions.conf file:
>>
>>
>> ;
>> ; Static extension configuration file, used by
>> ; the pbx_config module. This is where you configure all your
>> ; inbound and outbound calls in Asterisk.
>> ;
>> ; This configuration file is reloaded
>> ; - With the "extensions reload" command in the CLI
>> ; - With the "reload" command (that reloads everything) in the CLI
>>
>> ;
>> ; The "General" category is for certain variables.
>> ;
>> [general]
>> ;
>> ; If static is set to no, or omitted, then the pbx_config will rewrite
>> ; this file when extensions are modified.  Remember that all comments
>> ; made in the file will be lost when that happens.
>> ;
>> ; XXX Not yet implemented XXX
>> ;
>> static=yes
>> ;
>> ; if static=yes and writeprotect=no, you can save dialplan by
>> ; CLI command 'save dialplan' too
>> ;
>> writeprotect=no
>>
>> ; You can include other config files, use the #include command (without 
>> the
>> ';')
>> ; Note that this is different from the "include" command that includes
>> contexts within
>> ; other contexts. The #include command works in all asterisk 
>> configuration
>> files.
>> ;#include "filename.conf"
>>
>> ; The "Globals" category contains global variables that can be referenced
>> ; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
>> variable
>> ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
>> ;
>> [globals]
>> CONSOLE=Console/dsp    ; Console interface for demo
>> ;CONSOLE=Zap/1
>> ;CONSOLE=Phone/phone0
>> IAXINFO=guest     ; IAXtel username/password
>> ;IAXINFO=myuser:mypass
>> TRUNK=Zap/g2     ; Trunk interface
>> ;
>> ; Note the 'g2' in the TRUNK variable above. It specifies which group
>> (defined
>> ; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to 
>> use
>> in
>> ; the specified group. The four possible options are:
>> ;
>> ; g: select the lowest-numbered non-busy Zap channel (aka. ascending
>> sequential hunt group).
>> ; G: select the highest-numbered non-busy Zap channel (aka. descending
>> sequential hunt group).
>> ; r: use a round-robin search, starting at the next highest channel than
>> last time (aka. ascending rotary hunt group).
>> ; R: use a round-robin search, starting at the next lowest channel than 
>> last
>> time (aka. descending rotary hunt group).
>> ;
>> TRUNKMSD=1     ; MSD digits to strip (usually 1 or 0)
>> ;TRUNK=IAX2/user:pass at provider
>>
>> ;
>> ; Any category other than "General" and "Globals" represent
>> ; extension contexts, which are collections of extensions.
>> ;
>> ; Extension names may be numbers, letters, or combinations
>> ; thereof. If an extension name is prefixed by a '_'
>> ; character, it is interpreted as a pattern rather than a
>> ; literal.  In patterns, some characters have special meanings:
>> ;
>> ;   X - any digit from 0-9
>> ;   Z - any digit from 1-9
>> ;   N - any digit from 2-9
>> ;   [1235-9] - any digit in the brackets (in this example, 
>> 1,2,3,5,6,7,8,9)
>> ;   . - wildcard, matches anything remaining (e.g. _9011. matches
>> ; anything starting with 9011 excluding 9011 itself)
>> ;
>> ; For example the extension _NXXXXXX would match normal 7 digit dialings,
>> ; while _1NXXNXXXXXX would represent an area code plus phone number
>> ; preceeded by a one.
>> ;
>> ; Each step of an extension is ordered by priority, which must
>> ; always start with 1 to be considered a valid extension.
>> ;
>> ; Contexts contain several lines, one for each step of each
>> ; extension, which can take one of two forms as listed below,
>> ; with the first form being preferred.  One may include another
>> ; context in the current one as well, optionally with a
>> ; date and time.  Included contexts are included in the order
>> ; they are listed.
>> ;
>> ;[context]
>> ;exten => someexten,priority,application(arg1,arg2,...)
>> ;exten => someexten,priority,application,arg1|arg2...
>> ;
>> ; Timing list for includes is
>> ;
>> ;   <time range>|<days of week>|<days of month>|<months>
>> ;
>> ;include => daytime|9:00-17:00|mon-fri|*|*
>> ;
>> ; ignorepat can be used to instruct drivers to not cancel dialtone upon
>> ; receipt of a particular pattern.  The most commonly used example is
>> ; of course '9' like this:
>> ;
>> ;ignorepat => 9
>> ;
>> ; so that dialtone remains even after dialing a 9.
>> ;
>>
>> ;
>> ; Here are the entries you need to participate in the IAXTEL
>> ; call routing system.  Most IAXTEL numbers begin with 1-700, but
>> ; there are exceptions.  For more information, and to sign
>> ; up, please go to www.gnophone.com or www.iaxtel.com
>> ;
>> [iaxtel700]
>> exten => 
>> _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
>>
>> ;
>> ; The SWITCH statement permits a server to share the dialplain with
>> ; another server. Use with care: Reciprocal switch statements are not
>> ; allowed (e.g. both A -> B and B -> A), and the switched server needs
>> ; to be on-line or else dialing can be severly delayed.
>> ;
>> [iaxprovider]
>> ;switch => IAX2/user:[key]@myserver/mycontext
>>
>> [trunkint]
>> ;
>> ; International long distance through trunk
>> ;
>> exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>> exten => _9011.,2,Congestion
>>
>> [trunkld]
>> ;
>> ; Long distance context accessed through trunk
>> ;
>> exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>> exten => _91NXXNXXXXXX,2,Congestion
>>
>> [trunklocal]
>> ;
>> ; Local seven-digit dialing accessed through trunk interface
>> ;
>> exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>> exten => _9NXXXXXX,2,Congestion
>>
>> [trunktollfree]
>> ;
>> ; Long distance context accessed through trunk interface
>> ;
>> exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>> exten => _91800NXXXXXX,2,Congestion
>> exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>> exten => _91888NXXXXXX,2,Congestion
>> exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>> exten => _91877NXXXXXX,2,Congestion
>> exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
>> exten => _91866NXXXXXX,2,Congestion
>>
>> [international]
>> ;
>> ; Master context for international long distance
>> ;
>> ignorepat => 9
>> include => longdistance
>> include => trunkint
>>
>> [longdistance]
>> ;
>> ; Master context for long distance
>> ;
>> ignorepat => 9
>> include => local
>> include => trunkld
>>
>> [local]
>> ;
>> ; Master context for local, toll-free, and iaxtel calls only
>> ;
>> ignorepat => 9
>> include => default
>> include => parkedcalls
>> include => trunklocal
>> include => iaxtel700
>> include => trunktollfree
>> include => iaxprovider
>> ;
>> ; You can use an alternative switch type as well, to resolve
>> ; extensions that are not known here, for example with remote
>> ; IAX switching you transparently get access to the remote
>> ; Asterisk PBX
>> ;
>> ; switch => IAX2/user:password at bigserver/local
>>
>> [macro-stdexten];
>> ;
>> ; Standard extension macro:
>> ;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
>> ;   ${ARG2} - Device(s) to ring
>> ;
>> exten => s,1,Dial(${ARG2},20)     ; Ring the interface, 20 seconds 
>> maximum
>> exten => s,2,Goto(s-${DIALSTATUS},1)    ; Jump based on status
>> (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
>>
>> exten => s-NOANSWER,1,Voicemail(u${ARG1})  ; If unavailable, send to
>> voicemail w/ unavail announce
>> exten => s-NOANSWER,2,Goto(default,s,1)   ; If they press #, return to 
>> start
>>
>> exten => s-BUSY,1,Voicemail(b${ARG1})   ; If busy, send to voicemail w/ 
>> busy
>> announce
>> exten => s-BUSY,2,Goto(default,s,1)    ; If they press #, return to start
>>
>> exten => _s-.,1,Goto(s-NOANSWER,1)    ; Treat anything else as no answer
>>
>> exten => a,1,VoicemailMain(${ARG1})    ; If they press *, send the user 
>> into
>> VoicemailMain
>>
>> [demo]
>> ;
>> ; We start with what to do when a call first comes in.
>> ;
>> exten => s,1,Wait,1   ; Wait a second, just for fun
>> exten => s,2,Answer   ; Answer the line
>> exten => s,3,DigitTimeout,5  ; Set Digit Timeout to 5 seconds
>> exten => s,4,ResponseTimeout,10  ; Set Response Timeout to 10 seconds
>> exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
>> exten => s,6,BackGround(demo-instruct) ; Play some instructions
>>
>> exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
>> exten => 2,2,Goto(s,6)
>>
>> exten => 3,1,SetLanguage(fr)  ; Set language to french
>> exten => 3,2,Goto(s,5)   ; Start with the congratulations
>>
>> exten => 1000,1,Goto(default,s,1)
>> ;
>> ; We also create an example user, 1234, who is on the console and has
>> ; voicemail, etc.
>> ;
>> exten => 1234,1,Playback(transfer,skip)  ; "Please hold while..."
>>     ; (but skip if channel is not up)
>> exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
>>
>> exten => 1235,1,Voicemail(u1234)  ; Right to voicemail
>>
>> exten => 1236,1,Dial(Console/dsp)  ; Ring forever
>> exten => 1236,2,Voicemail(u1234)  ; Unless busy
>>
>> ;
>> ; # for when they're done with the demo
>> ;
>> exten => #,1,Playback(demo-thanks)  ; "Thanks for trying the demo"
>> exten => #,2,Hangup   ; Hang them up.
>>
>> ;
>> ; A timeout and "invalid extension rule"
>> ;
>> exten => t,1,Goto(#,1)   ; If they take too long, give up
>> exten => i,1,Playback(invalid)  ; "That's not valid, try again"
>>
>> ;
>> ; Create an extension, 500, for dialing the
>> ; Asterisk demo.
>> ;
>> exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
>> exten => 500,2,Dial(IAX2/guest at misery.digium.com/s at default) ; Call the
>> Asterisk demo
>> exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site
>> exten => 500,4,Goto(s,6)  ; Return to the start over message.
>>
>> ;
>> ; Create an extension, 600, for evaulating echo latency.
>> ;
>> exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
>> exten => 600,2,Echo   ; Do the echo test
>> exten => 600,3,Playback(demo-echodone) ; Let them know it's over
>> exten => 600,4,Goto(s,6)  ; Start over
>>
>> ;
>> ; Give voicemail at extension 8500
>> ;
>> exten => 8500,1,VoicemailMain
>> exten => 8500,2,Goto(s,6)
>> ;
>> ; Here's what a phone entry would look like (IXJ for example)
>> ;
>> ;exten => 1265,1,Dial(Phone/phone0,15)
>> ;exten => 1265,2,Goto(s,5)
>>
>> ;[mainmenu]
>> ;
>> ; Example "main menu" context with submenu
>> ;
>> ;exten => s,1,Answer
>> ;exten => s,2,Background(thanks)  ; "Thanks for calling press 1 for 
>> sales, 2
>> for support, ..."
>> ;exten => 1,1,Goto(submenu,s,1)
>> ;exten => 2,1,Hangup
>> ;include => default
>> ;
>> ;[submenu]
>> ;exten => s,1,Ringing     ; Make them comfortable with 2 seconds of 
>> ringback
>> ;exten => s,2,Wait,2
>> ;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales
>> department.  Press 1 for steve, 2 for..."
>> ;exten => 1,1,Goto(default,steve,1)
>> ;exten => 2,1,Goto(default,mark,2)
>>
>> [default]
>> ;
>> ; By default we include the demo.  In a production system, you
>> ; probably don't want to have the demo there.
>> ;
>> include => demo
>>
>> ;
>> ; Extensions like the two below can be used for FWD, Nikotel, sipgate 
>> etc.
>> ; Note that you must have a [sipprovider] section in sip.conf whereas
>> ; the otherprovider.net example does not require such a peer definition
>> ;
>> ;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
>> ;exten => 
>> _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
>>
>> ; Real extensions would go here. Generally you want real extensions to be 
>> 4
>> or 5
>> ; digits long (although there is no such requirement) and start with a
>> single
>> ; digit that is fairly large (like 6 or 7) so that you have plenty of 
>> room
>> to
>> ; overlap extensions and menu options without conflict.  You can alias 
>> them
>> with
>> ; names, too and use global variables
>>
>> ;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1 ; Channel hints for 
>> presence
>> ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
>> ;exten => 6245,1,Dial(${HINT},20,rtT)  ; Use hint as listed
>> ;exten => 6361,1,Dial(IAX2/JaneDoe,,rm)  ; ring without time limit
>> ;exten => 6389,1,Dial(MGCP/aaln/1 at 192.168.0.14)
>> ;exten => 6394,1,Dial(Local/6275/n)  ; this will dial ${MARK}
>>
>> ;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is
>> something like Zap/2
>> ;exten => mark,1,Goto(6275|1)   ; alias mark to 6275
>> ;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
>> ;exten => wil,1,Goto(6236|1)
>> ;
>> ; Some other handy things are an extension for checking voicemail via
>> ; voicemailmain
>> ;
>> ;exten => 8500,1,VoicemailMain
>> ;exten => 8500,2,Hangup
>> ;
>> ; Or a conference room (you'll need to edit meetme.conf to enable this 
>> room)
>> ;
>> ;exten => 8600,1,Meetme(1234)
>> ;
>> ; Or playing an announcement to the called party, as soon it answers
>> ;
>> ;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
>> ;
>> ; For more information on applications, just type "show applications" at
>> your
>> ; friendly Asterisk CLI prompt.
>> ;
>> ; 'show application <command>' will show details of how you
>> ; use that particular application in this file, the dial plan.
>> ;
>>
>>
>>
>>
>> ----- Original Message -----
>> From: "dbruce" <dbruce at bananatel.ca>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> <asterisk-users at lists.digium.com>
>> Sent: Sunday, July 24, 2005 8:39 PM
>> Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
>>
>>
>> > Marc: My answer is not incorrect... it is incomplete.
>> >
>> > The OP stipulated 2 extensions 200 and 202... and provided a sip debug
>> > indicating a call from 200 to 777.
>> >
>> > I pointed out the obvious.
>> >
>> > If the OP is dialing 202 on the phone, and the phone is dialing 777, 
>> > then
>> > he
>> > needs to look at the dialplan configuration of the phone. If he is 
>> > dialing
>> > 777 on the phone and expecting to reach 202, then he will need to have
>> > translations in the asterisk dialplan. But, the question was "what 
>> > should
>> > I
>> > be looking at?"... Using just the information provided, and the fact 
>> > that
>> > he
>> > is new to asterisk... without any further information... the first 
>> > thing
>> > he
>> > should be looking at is why the phone is trying to reach 777 when he 
>> > wants
>> > to reach 202... Many new users do not realize the complexity of the SIP
>> > protocol, and only really look at the trace in a general manner... 
>> > such
>> > as:
>> > INVITE
>> > 407 Proxy Authentication Required
>> > ACK
>> > INVITE
>> > 404 Not Found
>> > ACK
>> >
>> > The idea was to provide a clue... not to provide a complete working
>> > dialplan
>> > and phone configuration. Providing new users with "the complete 
>> > package"
>> > is
>> > a dis-service to them. They will only learn from thier mistakes and
>> > experiences.. providing clues allows them to expand their experience 
>> > and
>> > build their confidence... It requires them to look at the details and
>> > learn
>> > to analyse them.
>> >
>> > Regards,
>> > Derek
>> >
>> >
>> > ----- Original Message -----
>> > From: "Marc Storck" <marc.storck at msnetworks.lu>
>> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> > <asterisk-users at lists.digium.com>
>> > Sent: Sunday, July 24, 2005 12:53 PM
>> > Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
>> >
>> >
>> >> Derek: you reply is uncorrect. If Angus has the extension 777 in his
>> >> dialplan/extensions.conf which will dial 202. The name of the peer has
>> >> absolutely nothing to do with which number/name he would have to dial.
>> >> Without dialplan he will be unable to call any extension even 202, as
>> >> 202 is only the name of the peer.
>> >>
>> >> Angus: please paste your extensions.conf to pastebin.ca
>> >>
>> >> Regards,
>> >>
>> >> Marc
>> >>
>> >> dbruce wrote:
>> >> > It appears from the debug that extension 200 is trying to call 777, 
>> >> > not
>> >> > 202. Your Asterisk server can't find an extension 777 and returns 
>> >> > "404
>> >> > not found". That will explain why you can't call extension 777 from
>> >> > extension 200. If you want to call extension 202, you will need to 
>> >> > dial
>> >> > 202 on extension 200, not 777.
>> >> >
>> >> > Regards,
>> >> > Derek
>> >> >
>> >> >
>> >> >     ----- Original Message -----
>> >> >     *From:* Angus Comber <mailto:angus at iteloffice.com>
>> >> >     *To:* asterisk-users at lists.digium.com
>> >> >     <mailto:asterisk-users at lists.digium.com>
>> >> >     *Sent:* Sunday, July 24, 2005 11:51 AM
>> >> >     *Subject:* [Asterisk-Users] Why can't sip/200 call sip/202
>> >> >
>> >> >     I have 2 sip accounts setup - 200 and 202.  If I do sip show 
>> >> > peers
>> >> > I
>> >> >     get:
>> >> >
>> >> >     sip show peers
>> >> >     Name/username    Host            Dyn Nat ACL Mask
>> >> >     Port     Status
>> >> >     202/202          192.168.0.6      D          255.255.255.255
>> >> >     5060     Unmonitored
>> >> >     201/201          (Unspecified)    D          255.255.255.255
>> >> >     5060     Unmonitored
>> >> >     200/200          192.168.0.3      D          255.255.255.255
>> >> >     5060     Unmonitored
>> >> >
>> >> >     200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream 
>> >> > BT100
>> >> >     IP phone.
>> >> >
>> >> >     relevant bit of sip.conf:
>> >> >
>> >> >     [200]
>> >> >     username=200
>> >> >     type=friend
>> >> >     secret=1234
>> >> >     port=5060
>> >> >     nat=never
>> >> >     dtmfmode=rfc2833
>> >> >     context=default
>> >> >     callerid="Angus Comber" <200>
>> >> >     host=dynamic
>> >> >     disallow=all
>> >> >     allow=ulaw
>> >> >     allow=alaw
>> >> >     allow=g723.1
>> >> >     allow=g729
>> >> >
>> >> >     [202]
>> >> >     username=202
>> >> >     type=friend
>> >> >     secret=1234
>> >> >     port=5060
>> >> >     nat=never
>> >> >     dtmfmode=rfc2833
>> >> >     context=default
>> >> >     callerid="Sam Comber" <202>
>> >> >     host=dynamic
>> >> >     disallow=all
>> >> >     allow=ulaw
>> >> >     allow=alaw
>> >> >     allow=g723.1
>> >> >     allow=g729
>> >> >
>> >> >
>> >> >     But whenever I try to dial between phones I get this:
>> >> >
>> >> >
>> >> >     Sip read:
>> >> >
>> >> >     0 headers, 0 lines
>> >> >
>> >> >
>> >> >     Sip read:
>> >> >     INVITE sip:777 at 192.168.0.13;user=phone SIP/2.0
>> >> >     Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
>> >> >     From: "Angus Comber"
>> >> >     <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
>> >> >     To: <sip:777 at 192.168.0.13;user=phone>
>> >> >     Contact: <sip:200 at 192.168.0.3;user=phone>
>> >> >     Supported: replaces, timer
>> >> >     Call-ID: 11e4ca07b25c9335 at 192.168.0.3
>> >> >     <mailto:11e4ca07b25c9335 at 192.168.0.3>
>> >> >     CSeq: 45925 INVITE
>> >> >     User-Agent: Grandstream GXP2000 1.0.1.9
>> >> >     Max-Forwards: 70
>> >> >     Allow:
>> >> >
>> > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
>> >> >     Content-Type: application/sdp
>> >> >     Content-Length: 258
>> >> >
>> >> >     v=0
>> >> >     o=200 8000 8000 IN IP4 192.168.0.3
>> >> >     s=SIP Call
>> >> >     c=IN IP4 192.168.0.3
>> >> >     t=0 0
>> >> >     m=audio 5004 RTP/AVP 18 0 8 101
>> >> >     a=sendrecv
>> >> >     a=rtpmap:18 G729/8000
>> >> >     a=rtpmap:0 PCMU/8000
>> >> >     a=rtpmap:8 PCMA/8000
>> >> >     a=ptime:20
>> >> >     a=rtpmap:101 telephone-event/8000
>> >> >     a=fmtp:101 0-11
>> >> >
>> >> >     13 headers, 13 lines
>> >> >     Using latest request as basis request
>> >> >     Sending to 192.168.0.3 : 5060 (non-NAT)
>> >> >     Reliably Transmitting (no NAT):
>> >> >     SIP/2.0 407 Proxy Authentication Required
>> >> >     Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
>> >> >     From: "Angus Comber"
>> >> >     <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
>> >> >     To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
>> >> >     Call-ID: 11e4ca07b25c9335 at 192.168.0.3
>> >> >     <mailto:11e4ca07b25c9335 at 192.168.0.3>
>> >> >     CSeq: 45925 INVITE
>> >> >     User-Agent: Asterisk PBX
>> >> >     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> >> >     Contact: <sip:777 at 192.168.0.13>
>> >> >     Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366"
>> >> >     Content-Length: 0
>> >> >
>> >> >
>> >> >      to 192.168.0.3:5060
>> >> >     Scheduling destruction of call '11e4ca07b25c9335 at 192.168.0.3'
>> >> >     <mailto:'11e4ca07b25c9335 at 192.168.0.3'> in 15000 ms
>> >> >     Found user '200'
>> >> >
>> >> >
>> >> >     Sip read:
>> >> >     ACK sip:777 at 192.168.0.13;user=phone SIP/2.0
>> >> >     Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
>> >> >     From: "Angus Comber"
>> >> >     <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
>> >> >     To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
>> >> >     Contact: <sip:200 at 192.168.0.3;user=phone>
>> >> >     Call-ID: 11e4ca07b25c9335 at 192.168.0.3
>> >> >     <mailto:11e4ca07b25c9335 at 192.168.0.3>
>> >> >     CSeq: 45925 ACK
>> >> >     User-Agent: Grandstream GXP2000 1.0.1.9
>> >> >     Max-Forwards: 70
>> >> >     Allow:
>> >> >
>> > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
>> >> >     Content-Length: 0
>> >> >
>> >> >
>> >> >     11 headers, 0 lines
>> >> >
>> >> >
>> >> >     Sip read:
>> >> >     INVITE sip:777 at 192.168.0.13;user=phone SIP/2.0
>> >> >     Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
>> >> >     From: "Angus Comber"
>> >> >     <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
>> >> >     To: <sip:777 at 192.168.0.13;user=phone>
>> >> >     Contact: <sip:200 at 192.168.0.3;user=phone>
>> >> >     Supported: replaces, timer
>> >> >     Proxy-Authorization: Digest username="200", realm="asterisk",
>> >> >     algorithm=MD5, uri="sip:777 at 192.168.0.13;user=phone",
>> >> >     nonce="0c555366", response="ee6088fb4e50da5fe412913ae40dd45c"
>> >> >     Call-ID: 11e4ca07b25c9335 at 192.168.0.3
>> >> >     <mailto:11e4ca07b25c9335 at 192.168.0.3>
>> >> >     CSeq: 45926 INVITE
>> >> >     User-Agent: Grandstream GXP2000 1.0.1.9
>> >> >     Max-Forwards: 70
>> >> >     Allow:
>> >> >
>> > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
>> >> >     Content-Type: application/sdp
>> >> >     Content-Length: 258
>> >> >
>> >> >     v=0
>> >> >     o=200 8000 8001 IN IP4 192.168.0.3
>> >> >     s=SIP Call
>> >> >     c=IN IP4 192.168.0.3
>> >> >     t=0 0
>> >> >     m=audio 5004 RTP/AVP 18 0 8 101
>> >> >     a=sendrecv
>> >> >     a=rtpmap:18 G729/8000
>> >> >     a=rtpmap:0 PCMU/8000
>> >> >     a=rtpmap:8 PCMA/8000
>> >> >     a=ptime:20
>> >> >     a=rtpmap:101 telephone-event/8000
>> >> >     a=fmtp:101 0-11
>> >> >
>> >> >     14 headers, 13 lines
>> >> >     Using latest request as basis request
>> >> >     Sending to 192.168.0.3 : 5060 (non-NAT)
>> >> >     Found user '200'
>> >> >     Found RTP audio format 18
>> >> >     Found RTP audio format 0
>> >> >     Found RTP audio format 8
>> >> >     Found RTP audio format 101
>> >> >     Peer audio RTP is at port 192.168.0.3:5004
>> >> >     Found description format G729
>> >> >     Found description format PCMU
>> >> >     Found description format PCMA
>> >> >     Found description format telephone-event
>> >> >     Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - 
>> >> > audio=0x10c
>> >> >     (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c
>> > (ulaw|alaw|g729)
>> >> >     Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723),
>> >> > combined
>> >> >     - 0x1 (g723)
>> >> >     Looking for 777 in default
>> >> >     Reliably Transmitting (no NAT):
>> >> >     SIP/2.0 404 Not Found
>> >> >     Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
>> >> >     From: "Angus Comber"
>> >> >     <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
>> >> >     To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
>> >> >     Call-ID: 11e4ca07b25c9335 at 192.168.0.3
>> >> >     <mailto:11e4ca07b25c9335 at 192.168.0.3>
>> >> >     CSeq: 45926 INVITE
>> >> >     User-Agent: Asterisk PBX
>> >> >     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> >> >     Contact: <sip:777 at 192.168.0.13>
>> >> >     Content-Length: 0
>> >> >
>> >> >
>> >> >      to 192.168.0.3:5060
>> >> >
>> >> >
>> >> >     Sip read:
>> >> >     ACK sip:777 at 192.168.0.13;user=phone SIP/2.0
>> >> >     Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
>> >> >     From: "Angus Comber"
>> >> >     <sip:200 at 192.168.0.13;user=phone>;tag=a1afaf4fdb0ac845
>> >> >     To: <sip:777 at 192.168.0.13;user=phone>;tag=as668982be
>> >> >     Contact: <sip:200 at 192.168.0.3;user=phone>
>> >> >     Proxy-Authorization: Digest username="200", realm="asterisk",
>> >> >     algorithm=MD5, uri="sip:777 at 192.168.0.13;user=phone",
>> >> >     nonce="0c555366", response="7fcb1024a81b3ea3bcc56baeca4bac3e"
>> >> >     Call-ID: 11e4ca07b25c9335 at 192.168.0.3
>> >> >     <mailto:11e4ca07b25c9335 at 192.168.0.3>
>> >> >     CSeq: 45926 ACK
>> >> >     User-Agent: Grandstream GXP2000 1.0.1.9
>> >> >     Max-Forwards: 70
>> >> >     Allow:
>> >> >
>> > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
>> >> >     Content-Length: 0
>> >> >
>> >> >
>> >> >     12 headers, 0 lines
>> >> >     Destroying call '11e4ca07b25c9335 at 192.168.0.3'
>> >> >     <mailto:'11e4ca07b25c9335 at 192.168.0.3'>
>> >> >
>> >> >
>> >> >     How can I troubleshoot?  What should I be looking at?
>> >> >
>> >> >     Angus
>> >> >
>> >> >
>> >>
>>
>> 
>> >   ------------------------------------------------------------------------
>> >> >
>> >> >     _______________________________________________
>> >> >     Asterisk-Users mailing list
>> >> >     Asterisk-Users at lists.digium.com
>> >> >     http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> >     To UNSUBSCRIBE or update options visit:
>> >> >        http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> >
>> >> > _______________________________________________
>> >> > Asterisk-Users mailing list
>> >> > Asterisk-Users at lists.digium.com
>> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> > To UNSUBSCRIBE or update options visit:
>> >> >    http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >> --
>> >> CTO                            Marc Storck
>> >> MS Networks SA                 mstorck at msnetworks.lu
>> >> IT Service Provider            http://www.msnetworks.lu
>> >> 15, route d'Esch               Phone: +352 2727 3030
>> >> L-4450 Belvaux                 Fax:   +352 2727 3060
>> >>
>> >> --------------- MS Networks powered service ---------------
>> >> http://www.LuxAdmin.com       Hosting and housing solutions
>> >> -----------------------------------------------------------
>> >>
>> >> _______________________________________________
>> >> Asterisk-Users mailing list
>> >> Asterisk-Users at lists.digium.com
>> >> http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> To UNSUBSCRIBE or update options visit:
>> >>    http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> > _______________________________________________
>> > Asterisk-Users mailing list
>> > Asterisk-Users at lists.digium.com
>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> -- 
> regards,
>
> Mark P. Edwards
> FWD: 667917
> _______________________________________________
> 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
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> 





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