[Asterisk-Users] error 488 - Not Acceptable Here

Eric Wieling eric at fnords.org
Wed Apr 7 13:53:42 MST 2004


Take out the allow=all in your sip.conf and put in allow= for the codec
you want to use and disallow=all.

On Wed, 2004-04-07 at 15:18, Roger wrote:
> I have a setup of 3 Cisco 7940 running Sip image 6.3.  All these phone 
> are registered by the below information
> 
> *CLI> sip show peers
> Name/username    Host                 Mask             Port     Status
> 2002/2002        192.168.22.199  (D)  255.255.255.255  5060     Unmonitored
> 2001/2001        192.168.22.200  (D)  255.255.255.255  5060     Unmonitored
> 2000/2000        192.168.22.198  (D)  255.255.255.255  5060     Unmonitored
> 
> *CLI> sip show users
> Username         Secret           Authen           Def.Context      A/C
> 2002             cisco            md5,plaintext    demo             No
> 2001             cisco            md5,plaintext    demo             No
> 2000             cisco            md5,plaintext    demo             No
> 
> 
> All 3 phones and the asterisk box are on the 192.168.22.0/24 subnet.  
> I've attached my sip.conf and extensions.conf file for review...
> 
> When I start the server and a phone dials another phone I get the below 
> answer. 
> 
> *CLI>     -- Executing Dial("SIP/2001-0bb5", "SIP/2002|30|tr") in new stack
>     -- Called 2002
>     -- Got SIP response 488 "Not Acceptable Here" back from 192.168.22.199
>   == No one is available to answer at this time
>     -- Timeout on SIP/2001-0bb5
> 
> I *believe* the sip response might be from the phone itself - and not a 
> asterisk misconfig.  I'm just wanting a second pair of eyes.
> 
> I put in
> 
> canreinvite=no
> 
> for each phone profile as people have said this is needed for buggy 
> Cisco phones.
> 
> 
> ______________________________________________________________________
> ;
> ; SIP Configuration for Asterisk
> ;
> ; Syntax for specifying a SIP device in extensions.conf is
> ; SIP/devicename where devicename is defined in a section below.
> ;
> ; You may also use
> ; SIP/username at domain to call any SIP user on the Internet
> ; (Don't forget to enable DNS SRV records if you want to use this)
> ;
> ; If you define a SIP proxy as a peer below, you may call
> ; SIP/proxyhostname/user or SIP/user at proxyhostname
> ; where the proxyhostname is defined in a section below
> ;
> ; Useful CLI commands to check peers/users:
> ;   sip show peers              Show all SIP peers (including friends)
> ;   sip show users              Show all SIP users (including friends)
> ;   sip show registry           Show status of hosts we register with
> ;
> ;   sip debug                   Show all SIP messages
> ;
> 
> [general]
> 
> port = 5060           ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0	; Address to bind to (all addresses on machine)
> ;bindaddr = 192.168.22.254    ; Address to bind to (all addresses on machine)
> allow=all             ; Allow all codecs
> context = bogon-calls ; Send SIP callers that we don't know about here
> tos = lowdelay	      ; can be lowdelay, throughput, reliability, mincost
> 
> [2000]
> type=friend           ; This device takes and makes calls
> username=2000         ; Username on device
> secret=cisco		; Password for device
> ;host=192.168.22.1   ; This host is not on the same IP addr every time
> host=dynamic
> context=demo		; Inbound calls from this host go here
> mailbox=100           ; Activate the message waiting light if this
> canreinvite=no
>                       ; voicemailbox has messages in it
> 
> [2001]                ; Duplicate of 2000, except with different auth data
> type=friend
> username=2001
> secret=cisco
> host=dynamic
> ;host=192.168.22.2
> context=demo
> mailbox=101
> canreinvite=no
> 
> [2002]                ; Duplicate of 2000, except with different auth data
> type=friend
> username=2002
> secret=cisco
> ;host=192.168.22.3
> host=dynamic
> context=demo
> mailbox=102
> canreinvite=no
> 
> ______________________________________________________________________
> ;
> ; Static extension configuration file, used by
> ; the pbx_config module. This is where you configure all your 
> ; inbound and outbound calls in Asterisk. 
> ; 
> [incoming]
> exten => s,1,Echo ;for testing the connection
> ;exten => s,1,Playback,demo-thanks ;for playing a file
> ;
> ; 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
> 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.
> ;
> ; 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 => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
> 
> [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,Voicemail(u${ARG1})				; If unavailable, send to voicemail w/ unavail announce
> exten => s,3,Goto(default,s,1)					; If they press #, return to start
> exten => s,102,Voicemail(b${ARG1})				; If busy, send to voicemail w/ busy announce
> exten => s,103,Goto(default,s,1)				; If they press #, return to start
> 
> 
> [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
> 
> exten => 2000,1,Dial(SIP/2000,30,tr)
> 
> exten => 2001,1,Dial(SIP/2001,30,tr)
> 
> exten => 2002,1,Dial(SIP/2002,30,tr)
> 
> ;
> ; # 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
> 
> 
> ; 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 => 6275,1,Macro(stdexten,6275,${MARK})			; assuming ${MARK} is something like Zap/2
> ;exten => mark,1,Goto(6275|1)						; alias mark to 6275
> ;exten => 6236,1,Macro(stdexten,6236,${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. 
> ;
-- 
          Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."




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