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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>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
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> 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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