The extensions.conf file you provided looks suspiciously like the asterisk
configs/extensions.conf.sample file.
Did you create a dialplan for your specific configuration or did you just
copy the sample file?
----- Original Message -----
From: "Angus Comber" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Sent: Sunday, July 24, 2005 2:50 PM
Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202
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:[EMAIL PROTECTED]
;
; 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/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
;
; The SWITCH statement permits a server to share the dialplain with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[EMAIL PROTECTED]/mycontext
[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Dial(${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:[EMAIL PROTECTED]/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/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; 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:[EMAIL PROTECTED],,r)
;exten =>
_42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],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/[EMAIL PROTECTED])
;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" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@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" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@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:[EMAIL PROTECTED]>
>> > *To:* asterisk-users@lists.digium.com
>> > <mailto:asterisk-users@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:[EMAIL PROTECTED];user=phone SIP/2.0
>> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
>> > From: "Angus Comber"
>> > <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845
>> > To: <sip:[EMAIL PROTECTED];user=phone>
>> > Contact: <sip:[EMAIL PROTECTED];user=phone>
>> > Supported: replaces, timer
>> > Call-ID: [EMAIL PROTECTED]
>> > <mailto:[EMAIL PROTECTED]>
>> > 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:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845
>> > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be
>> > Call-ID: [EMAIL PROTECTED]
>> > <mailto:[EMAIL PROTECTED]>
>> > CSeq: 45925 INVITE
>> > User-Agent: Asterisk PBX
>> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> > Contact: <sip:[EMAIL PROTECTED]>
>> > Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366"
>> > Content-Length: 0
>> >
>> >
>> > to 192.168.0.3:5060
>> > Scheduling destruction of call '[EMAIL PROTECTED]'
>> > <mailto:'[EMAIL PROTECTED]'> in 15000 ms
>> > Found user '200'
>> >
>> >
>> > Sip read:
>> > ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
>> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1
>> > From: "Angus Comber"
>> > <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845
>> > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be
>> > Contact: <sip:[EMAIL PROTECTED];user=phone>
>> > Call-ID: [EMAIL PROTECTED]
>> > <mailto:[EMAIL PROTECTED]>
>> > 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:[EMAIL PROTECTED];user=phone SIP/2.0
>> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
>> > From: "Angus Comber"
>> > <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845
>> > To: <sip:[EMAIL PROTECTED];user=phone>
>> > Contact: <sip:[EMAIL PROTECTED];user=phone>
>> > Supported: replaces, timer
>> > Proxy-Authorization: Digest username="200", realm="asterisk",
>> > algorithm=MD5, uri="sip:[EMAIL PROTECTED];user=phone",
>> > nonce="0c555366", response="ee6088fb4e50da5fe412913ae40dd45c"
>> > Call-ID: [EMAIL PROTECTED]
>> > <mailto:[EMAIL PROTECTED]>
>> > 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:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845
>> > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be
>> > Call-ID: [EMAIL PROTECTED]
>> > <mailto:[EMAIL PROTECTED]>
>> > CSeq: 45926 INVITE
>> > User-Agent: Asterisk PBX
>> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> > Contact: <sip:[EMAIL PROTECTED]>
>> > Content-Length: 0
>> >
>> >
>> > to 192.168.0.3:5060
>> >
>> >
>> > Sip read:
>> > ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
>> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304
>> > From: "Angus Comber"
>> > <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845
>> > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be
>> > Contact: <sip:[EMAIL PROTECTED];user=phone>
>> > Proxy-Authorization: Digest username="200", realm="asterisk",
>> > algorithm=MD5, uri="sip:[EMAIL PROTECTED];user=phone",
>> > nonce="0c555366", response="7fcb1024a81b3ea3bcc56baeca4bac3e"
>> > Call-ID: [EMAIL PROTECTED]
>> > <mailto:[EMAIL PROTECTED]>
>> > 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 '[EMAIL PROTECTED]'
>> > <mailto:'[EMAIL PROTECTED]'>
>> >
>> >
>> > How can I troubleshoot? What should I be looking at?
>> >
>> > Angus
>> >
>> >
>>
------------------------------------------------------------------------
>> >
>> > _______________________________________________
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>> > Asterisk-Users@lists.digium.com
>> > http://lists.digium.com/mailman/listinfo/asterisk-users
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>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> >
>>
------------------------------------------------------------------------
>> >
>> > _______________________________________________
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>> > Asterisk-Users@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 [EMAIL PROTECTED]
>> 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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>
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