[Asterisk-Users] Why can't sip/200 call sip/202
Angus Comber
angus at iteloffice.com
Sun Jul 24 15:30:11 MST 2005
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
> >
>
>
> _______________________________________________
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>
--
regards,
Mark P. Edwards
FWD: 667917
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