Rejected because extension not found in context


I have this in sip.conf:

[basic-options](!)                ; a template

[natted-phone](!,basic-options)   ; another template inheriting basic-options

[public-phone](!,basic-options)   ; another template inheriting basic-options

[my-codecs](!)                    ; a template for my preferred codecs

[ulaw-phone](!)                   ; and another one for ulaw-only

; and finally instantiate a few phones
        secret = 2133
        secret = 2134
        secret = 2136

in extensions.conf i have this:[code]

exten => _2XXX,1,Dial(SIP/${EXTEN},20,tT)
exten => _2XXX,2,Hangup()
include => parkedcalls

The Extension 2133 is a Cisco SPA504G (Register Failed).
The Extension 2134 is my JavaSE Application (Register Successful).
The Extension 2136 is Peers Java SIP Client (Register Successful).

My problem, besides that Cisco SPA504G Resister is Failed, is that when I make a call I get the following warning in the CLI:

[May  3 16:27:47] NOTICE[2125]: chan_sip.c:21485 handle_request_invite: Call from '2136' to extension '2134' rejected because extension not found in context 'internal'.

Please, could someone help me.

Best regards

Asterisk built by root @ localhost.localdomain on a x86_64 running Linux on 2011-05-01 12:27:06 UTC
Linux localhost.localdomain #1 SMP Fri Apr 22 16:01:29 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux

localhost*CLI> sip show peers
Name/username              Host                                    Dyn Forcerport ACL Port     Status     
2133                       (Unspecified)                            D          0        Unmonitored 
2134/2134                                     D          6061     Unmonitored 
2136/2136                                     D          6060     Unmonitored 
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 1 offline]

dialplan show internal
[ Context 'internal' created by 'pbx_config' ]
  '_2XXX' =>        1. Dial(SIP/${EXTEN},20,tT)                   [pbx_config]
                    2. Hangup()                                   [pbx_config]

-= 1 extension (2 priorities) in 1 context. =-

is the cisco phone local or remote?
You dont show the portion of sip.conf that relates to the sip phone???

Thanks riddlebox for your reply,

do not quite understand the question but this is how I have the SPA504:

A – Cisco SPA504
B – Asterisk 1.8
C – Huaweii Router

A and B are connected to C.

I guess it’s local

In sip.conf i have this for Cisco SPA:

[2133](public-phone,my-codecs) secret = 2133

Cisco phone show the extenion 2133 in display, but ne orange flashing lights would have to be green when it register the extension.

Sorry for my english.

Best regards

Looks like a bug…

  1. Try not to use templates within sip.conf
  2. Try to use 1.4.X branch with this config files

If somethig of this will help, go to bugtracker )

this same configuration worked fine in Asterisk

I’m installing that version to see if it works

Thank you not work for me.

make uninstall && make uninstall-all

wget Asterisk && ./configure && make menuselect && make && make install && make samples

This is my full extensions.conf:

[code]; extensions.conf - the Asterisk dial plan
; 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 “dialplan reload” command in the CLI
; - With the “reload” command (that reloads everything) in the CLI

; The “General” category is for certain variables.
; 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
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command “dialplan save” too
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk’s best guess. This is the default.
; If autofallthrough is not set, then if an extension runs out of
; things to do, Asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
; If extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses
; a Trie to find the best matching pattern is used. In dialplans
; with more than about 20-40 extensions in a single context, this
; new algorithm can provide a noticeable speedup.
; With 50 extensions, the speedup is 1.32x
; with 88 extensions, the speedup is 2.23x
; with 138 extensions, the speedup is 3.44x
; with 238 extensions, the speedup is 5.8x
; with 438 extensions, the speedup is 10.4x
; With 1000 extensions, the speedup is ~25x
; with 10,000 extensions, the speedup is 374x
; Basically, the new algorithm provides a flat response
; time, no matter the number of extensions.
; By default, the old pattern matcher is used.
; ****This is a new feature! *********************
; The new pattern matcher is for the brave, the bold, and
; the desperate. If you have large dialplans (more than about 50 extensions
; in a context), and/or high call volume, you might consider setting
; this value to “yes” !!
; Please, if you try this out, and are forced to return to the
; old pattern matcher, please report your reasons in a bug report
; on We have made good progress in providing
; something compatible with the old matcher; help us finish the job!
; This value can be switched at runtime using the cli command “dialplan set extenpatternmatchnew true”
; or “dialplan set extenpatternmatchnew false”, so you can experiment to your hearts content.
; If clearglobalvars is set, global variables will be cleared
; and reparsed on a dialplan reload, or Asterisk reload.
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one of its included files, will remain set to the previous value.
; NOTE: A complication sets in, if you put your global variables into
; the AEL file, instead of the extensions.conf file. With clearglobalvars
; set, a “reload” will often leave the globals vars cleared, because it
; is not unusual to have extensions.conf (which will have no globals)
; load after the extensions.ael file (where the global vars are stored).
; So, with “reload” in this particular situation, first the AEL file will
; clear and then set all the global vars, then, later, when the extensions.conf
; file is loaded, the global vars are all cleared, and then not set, because
; they are not stored in the extensions.conf file.
; If priorityjumping is set to ‘yes’, then applications that support
; ‘jumping’ to a different priority based on the result of their operations
; will do so (this is backwards compatible behavior with pre-1.2 releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a ‘j’ option in their arguments.
; User context is where entries from users.conf are registered. The
; default value is ‘default’
; 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”
;#include <filename.conf>
;#include filename.conf
; You can execute a program or script that produces config files, and they
; will be inserted where you insert the #exec command. The #exec command
; works on all asterisk configuration files. However, you will need to
; activate them within asterisk.conf with the “execincludes” option. They
; are otherwise considered a security risk.
;#exec /opt/bin/
;#exec /opt/bin/ --foo=“bar”
;#exec </opt/bin/ --foo=“bar”>
;#exec “/opt/bin/ --foo=“bar””

; The “Globals” category contains global variables that can be referenced
; in the dialplan with the GLOBAL dialplan function:
; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
; Unix/Linux environmental variables can be reached with the ENV dialplan
; function: ${ENV(VARIABLE)}
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=DAHDI/G2 ; Trunk interface
; Note the ‘G2’ in the TRUNK variable above. It specifies which group (defined
; in chan_dahdi.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 DAHDI channel
; (aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy DAHDI 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)

; ; domain to send on outbound
; freenum calls (uses outbound-freenum
; context)

; If you load any other extension configuration engine, such as,
; your global variables may be overridden by that file. Please take care to
; use only one location to set global variables, and you will likely save
; yourself a ton of grief.
; 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)
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are possible
; For example, the extension _NXXXXXX would match normal 7 digit dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceded by a one.
; Each step of an extension is ordered by priority, which must always start
; with 1 to be considered a valid extension. The priority “next” or “n” means
; the previous priority plus one, regardless of whether the previous priority
; was associated with the current extension or not. The priority “same” or “s”
; means the same as the previously specified priority, again regardless of
; whether the previous entry was for the same extension. Priorities may be
; immediately followed by a plus sign and another integer to add that amount
; (most useful with ‘s’ or ‘n’). Priorities may then also have an alias, or
; label, in parentheses after their name which can be used in goto situations.
; Contexts contain several lines, one for each step of each extension. 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.
; Switches may also be included within a context. The order of matching within
; a context is always exact extensions, pattern match extensions, includes, and
; switches. Includes are always processed depth-first. So for example, if you
; would like a switch “A” to match before context “B”, simply put switch “A” in
; an included context “C”, where “C” is included in your original context
; before “B”.
;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,…)
; Timing list for includes is
; ,,,[,]
; Note that ranges may be specified to wrap around the ends. Also, minutes are
; fine-grained only down to the closest even minute.
;include => daytime,9:00-17:00,mon-fri,,
;include => weekend,,sat-sun,,*
;include => weeknights,17:02-8:58,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. Please note that ignorepat
; only works with channels which receive dialtone from the PBX, such as DAHDI,
; Phone, and VPB. Other channels, such as SIP and MGCP, which generate their
; own dialtone and converse with the PBX only after a number is complete, are
; generally unaffected by ignorepat (unless DISA or another method is used to
; generate a dialtone after answering the channel).

; Sample entries for extensions.conf
;include => stdexten
; List canonical entries here
;exten => 12564286000,1,Gosub(6000,stdexten(IAX2/foo))
;exten => 12564286000,n,Goto(default,s,1) ; exited Voicemail
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})

; If you are an ITSP or Reseller, list your customers here.
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)

; If you are freely delivering calls to the PSTN, list them here
;exten => _1256428XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(DAHDI/G2/${EXTEN:7}) ; Ditto for 256-325

; Context to put your dundi IAX2 or SIP user in for
; full access
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

; Just a wrapper for the switch
switch => DUNDi/e164

; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don’t have one.
include => dundi-e164-local
include => dundi-e164-switch
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
; ARG1 is the extension to Dial
; Extension “s” is not a wildcard extension that matches “anything”.
; In macros, it is the start extension. In most other cases,
; you have to goto “s” to execute that extension.
; For wildcard matches, see above - all pattern matches start with
; an underscore.
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

; 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 or
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}${EXTEN:1}@iaxtel)

; The SWITCH statement permits a server to share the dialplan 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.
;switch => IAX2/user:[key]@myserver/mycontext

; International long distance through trunk
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})

; Long distance context accessed through trunk
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})

; Local seven-digit dialing accessed through trunk interface

; Long distance context accessed through trunk interface
exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

; Master context for international long distance
ignorepat => 9
include => longdistance
include => trunkint

; Master context for long distance
ignorepat => 9
include => local
include => trunkld

; Master context for local, toll-free, and iaxtel calls only
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider

;Include parkedcalls (or the context you define in features conf)
;to enable call parking.
include => parkedcalls
; 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@bigserver/local
; An “lswitch” is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
; lswitch => Loopback/12${EXTEN}@othercontext
; An “eswitch” is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
; eswitch => IAX2/context@${CURSERVER}

; The following two contexts are a template to enable the ability to dial
; ISN numbers. For more information about what an ISN number is, please see
; This is the dialing hook. use:
; include => outbound-freenum

; We’ll add more digits as needed. The purpose is to dial things
; like extension numbers at domains (ITAD number) so we’re matching
; on lengths of 1 through 6 prior to the separator (the asterisk [])
exten => _X
exten => _XXX!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXX
exten => _XXXXX!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXX
exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)

; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ; make sure the suffix is all digits as well
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != “${SUFFIX}”]?fn-CONGESTION,1)
; filter out bad characters per the document
same => n,Set(TIMEOUT(absolute)=10800)
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,1,}) ; perform our lookup with
same => n,GotoIf($["${isnresult}" != “”]?from)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = “”]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we’ll use it for our outbound dialing domain
same => n(dial),Dial(SIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)

exten => fn-BUSY,1,Busy()

exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()

; Standard trunk dial macro (hangs up on a dialstatus that should
; terminate call)
; ${ARG1} - What to dial
exten => s,1,Dial(${ARG1})
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Hangup
exten => _s-.,1,NoOp

; Standard extension subroutine:
; ${EXTEN} - Extension
; ${ARG1} - Device(s) to ring
; ${ARG2} - Optional context in Voicemail
; Note that the current version will drop through to the next priority in the
; case of their pressing ‘#’. This gives more flexibility in what do to next:
; you can prompt for a new extension, or drop the call, or send them to a
; general delivery mailbox, or…
; The use of the LOCAL() function is purely for convenience. Any variable
; initially declared as LOCAL() will disappear when the innermost Gosub context
; in which it was declared returns. Note also that you can declare a LOCAL()
; variable on top of an existing variable, and its value will revert to its
; previous value (before being declared as LOCAL()) upon Return.
exten => _X.,50000(stdexten),NoOp(Start stdexten)
exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
exten => _X.,n,Set(LOCAL(dev)=${ARG1})
exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
exten => _X.,n,Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})})
exten => _X.,n,Dial(${dev},20) ; Ring the interface, 20 seconds maximum
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start

exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,Return() ; If they press #, return to start

exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return()

; Standard extension subroutine:
; ${ARG1} - Extension
; ${ARG2} - Device(s) to ring
; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
; ${ARG5} - Context in voicemail (if empty, then “default”)
; See above note in stdexten about priority handling on exit.
exten => _X.,60000(stdPrivacyexten),NoOp(Start stdPrivacyexten)
exten => _X.,n,Set(LOCAL(ext)=${ARG1})
exten => _X.,n,Set(LOCAL(dev)=${ARG2})
exten => _X.,n,Set(LOCAL(dontcntx)=${ARG3})
exten => _X.,n,Set(LOCAL(tortcntx)=${ARG4})
exten => _X.,n,Set(LOCAL(cntx)=${ARG5})

exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? “@${cntx}” :: “”])
exten => _X.,n,Dial(${dev},20,p) ; Ring the interface, 20 seconds maximum, call screening
; option (or use P for databased call _X.creening)
exten => _X.,n,Goto(stdexten-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => stdexten-NOANSWER,1,Voicemail(${mbx},u) ; If unavailable, send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,NoOp(Finish stdPrivacyexten NOANSWER)
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start

exten => stdexten-BUSY,1,Voicemail(${mbx},b) ; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,NoOp(Finish stdPrivacyexten BUSY)
exten => stdexten-BUSY,n,Return() ; If they press #, return to start

exten => stdexten-DONTCALL,1,Goto(${dontcntx},s,1) ; Callee chose to send this call to a polite “Don’t call again” script.

exten => stdexten-TORTURE,1,Goto(${tortcntx},s,1) ; Callee chose to send this call to a telemarketer torture script.

exten => _stde[x]te[n]-.,1,Goto(stdexten-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${mbx}) ; If they press *, send the user into VoicemailMain
exten => a,n,Return

; Paging macro:
; Check to see if SIP device is in use and DO NOT PAGE if they are
; ${ARG1} - Device to page

exten => s,1,ChanIsAvail(${ARG1},s) ; s is for ANY call
exten => s,n,GoToIf($[${AVAILSTATUS} = “1”]?autoanswer:fail)
exten => s,n(autoanswer),Set(_ALERT_INFO=“RA”) ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the Grandstream, Snoms, and Others
exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!
exten => s,n,Dial(${ARG1})
exten => s,n(fail),Hangup

include => stdexten
; 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,n,Answer ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.

exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; 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,n,Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)}))
exten => 1234,n,Goto(default,s,1) ; exited Voicemail

exten => 1235,1,Voicemail(1234,u) ; Right to voicemail

exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(1234,b) ; Unless busy

; # for when they’re done with the demo
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,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,n,Dial(IAX2/ ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn’t connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.

; Create an extension, 600, for evaluating echo latency.
exten => 600,1,Playback(demo-echotest) ; Let them know what’s going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it’s over
exten => 600,n,Goto(s,6) ; Start over

; You can use the Macro Page to intercom a individual user
exten => 76245,1,Macro(page,SIP/Grandstream1)
; or if your peernames are the same as extensions
exten => _7XXX,1,Macro(page,SIP/${EXTEN})
; System Wide Page at extension 7999
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)

; Give voicemail at extension 8500
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
; Here’s what a phone entry would look like (IXJ for example)
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

; The page context calls up the page macro that sets variables needed for auto-answer
; It is in is own context to make calling it from the Page() application as simple as
; Local/{peername}@page
exten => _X.,1,Macro(page,SIP/${EXTEN})

; Example “main menu” context with submenu
;exten => s,1,Answer
;exten => s,n,Background(thanks) ; “Thanks for calling press 1 for sales, 2 for support, …”
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts) ; “Thanks for calling the sales department. Press 1 for steve, 2 for…”
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

; By default we include the demo. In a production system, you
; probably don’t want to have the demo there.
include => demo

; An extension like the one below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf
;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,r)

; 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(Joe Schmoe) ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
;exten => 6245,n,Voicemail(6245,u) ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup ; s+1, same as n
;exten => 6245,dial+101,Voicemail(6245,b) ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,rm) ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1@
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
;exten => 6391,1,Dial(JINGLE/ ;Dial via jingle using asterisk as the transport and calling mogorman.
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}

;exten => 6275,1,Gosub(${EXTEN},stdexten(${MARK}))
; assuming ${MARK} is something like DAHDI/2
;exten => 6275,n,Goto(default,s,1) ; exited Voicemail
;exten => mark,1,Goto(6275,1) ; alias mark to 6275
;exten => 6536,1,Gosub(${EXTEN},stdexten(${WIL}))
; Ditto for wil
;exten => 6536,n,Goto(default,s,1) ; exited Voicemail
;exten => wil,1,Goto(6236,1)

;If you want to subscribe to the status of a parking space, this is
;how you do it. Subscribe to extension 6600 in sip, and you will see
;the status of the first parking lot with this extensions’ help
;exten => 6600,hint,park:701@parkedcalls
;exten => 6600,1,noop
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;exten => 8500,1,VoicemailMain
;exten => 8500,n,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))

; example of a compartmentalized company called “acme”
; this is the context that your incoming IAX/SIP trunk dumps you in…
;exten => s,1,Wait(1)
;exten => s,n,Answer()
;exten => s,n(menu),Playback(acme/vm-brief-menu)
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
;exten => s,n(goodbye),Playback(vm-goodbye)
;exten => s,n(end),Hangup()
;include => acme-extens
;exten => i,1,Playback(vm-invalid)
;exten => i,n,Goto(s,exten) ; optionally, transfer to operator
;exten => t,1,Goto(s,goodbye)
; this is the context our internal SIP hardphones use (see sip.conf)
;exten => s,1,Answer()
;exten => s,n(exten),Background(vm-enter-num-to-call)
;exten => s,n,WaitExten(5)
;exten => s,n(goodbye),Playback(vm-goodbye)
;exten => s,n(end),Hangup()
;include => trunkint
;include => trunkld
;include => trunklocal
;include => acme-extens
; you can test what your system sounds like to outside callers by dialing this
;exten => 777,1,DISA(no-password,acme-incoming)
; grouping of acme’s extensions… never used directly, always included.
;include => stdexten
;exten => 111,1,Gosub(111,stdexten(SIP/pete_1,acme))
;exten => 111,n,Goto(s,exten)
;exten => 112,1,Gosub(112,stdexten(SIP/nancy_1,acme))
;exten => 112,n,Goto(s,end)
; end of acme example

; Time context: you can patch this in via the following.
; [acme-internal]
; …
; exten => 777,1,Gosub(time)
; exten => 777,n,Hangup()
; …
; include => time
; Note: if you’re geographically spread out, you can have SIP extensions
; specify their own local timezone in sip.conf as:
; [boi]
; type=friend
; context=acme-internal
; callerid=“Boise Ofc. <2083451111>”
; …
; ; use system-wide default timezone of MST7MDT
; [lws]
; type=friend
; context=acme-internal
; callerid=“Lewiston Ofc. <2087431111>”
; …
; setvar=timezone=PST8PDT
; “timezone” isn’t a ‘reserved’ name in any way, and other places where
; the timezone is significant (e.g. calls to “SayUnixTime()”, etc) will
; require modification as well. Note that voicemail.conf already has
; a mechanism for timezones.

exten => _X.,30000(time),NoOp(Time: ${EXTEN} ${timezone})
exten => _X.,n,Wait(0.25)
exten => _X.,n,Answer()
; the amount of delay is set for English; you may need to adjust this time
; for other languages if there’s no pause before the synchronizing beep.
exten => _X.,n,Set(FUTURETIME=$[${EPOCH} + 12])
exten => _X.,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
exten => _X.,n,SayPhonetic(z)
; use the timezone associated with the extension (sip only), or system-wide
; default if one hasn’t been set.
exten => _X.,n,SayUnixTime(${FUTURETIME},${timezone},HNS)
exten => _X.,n,Playback(spy-local)
exten => _X.,n,WaitUntil(${FUTURETIME})
exten => _X.,n,Playback(beep)
exten => _X.,n,Return()

; ANI context: use in the same way as “time” above

exten => _X.,40000(ani),NoOp(ANI: ${EXTEN})
exten => _X.,n,Wait(0.25)
exten => _X.,n,Answer()
exten => _X.,n,Playback(vm-from)
exten => _X.,n,SayDigits(${CALLERID(ani)})
exten => _X.,n,Wait(1.25)
exten => _X.,n,SayDigits(${CALLERID(ani)}) ; playback again in case of missed digit
exten => _X.,n,Return()

; For more information on applications, just type “core show applications” at your
; friendly Asterisk CLI prompt.
; "core show application " will show details of how you
; use that particular application in this file, the dial plan.
; “core show functions” will list all dialplan functions
; "core show function " will show you more information about
; one function. Remember that function names are UPPER CASE.

exten =>_8XXX,1,Dial(SIP/${EXTEN},20,tT)
exten =>_8XXX,2,Hangup()

This is my sip.conf:

; SIP Configuration example for Asterisk
; Note: Please read the security documentation for Asterisk in order to
; understand the risks of installing Asterisk with the sample
; configuration. If your Asterisk is installed on a public
; IP address connected to the Internet, you will want to learn
; about the various security settings BEFORE you start
; Asterisk.
; Especially note the following settings:
; - allowguest (default enabled)
; - permit/deny - IP address filters
; - contactpermit/contactdeny - IP address filters for registrations
; - context - Which set of services you offer various users
; SIP dial strings
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
; SIP/devicename
; SIP/username@domain (SIP uri)
; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
; SIP/devicename/extension
; SIP/devicename/extension/IPorHost
; SIP/username@domain//IPorHost
; Devicename
; devicename is defined as a peer in a section below.
; username@domain
; Call any SIP user on the Internet
; (Don’t forget to enable DNS SRV records if you want to use this)
; devicename/extension
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
; This syntax also works with ATA’s with FXO ports
; SIP/username[:password[:md5secret[]]]@host[:port]
; This form allows you to specify password or md5secret and authname
; without altering any authentication data in config.
; Examples:
; SIP/*98@mysipproxy
; SIP/
; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@
; IPorHost
; The next server for this call regardless of domain/peer
; All of these dial strings specify the SIP request URI.
; In addition, you can specify a specific To: header by adding an
; exclamation mark after the dial string, like
; SIP/sales@mysipproxy!
; A new feature for 1.8 allows one to specify a host or IP address to use
; when routing the call. This is typically used in tandem with func_srv if
; multiple methods of reaching the same domain exist. The host or IP address
; is specified after the third slash in the dialstring. Examples:
; SIP/devicename/extension/IPorHost
; SIP/username@domain//IPorHost
; CLI Commands
; -------------------------------------------------------------
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show registry Show status of hosts we register with
; sip set debug on Show all SIP messages
; sip reload Reload configuration file
; sip show settings Show the current channel configuration
;------- Naming devices ------------------------------------------------------
; When naming devices, make sure you understand how Asterisk matches calls
; that come in.
; 1. Asterisk checks the SIP From: address username and matches against
; names of devices with type=user
; The name is the text between square brackets [name]
; 2. Asterisk checks the From: addres and matches the list of devices
; with a type=peer
; 3. Asterisk checks the IP address (and port number) that the INVITE
; was sent from and matches against any devices with type=peer
; Don’t mix extensions with the names of the devices. Devices need a unique
; name. The device name is not used as phone numbers. Phone numbers are
; anything you declare as an extension in the dialplan (extensions.conf).
; When setting up trunks, make sure there’s no risk that any From: username
; (caller ID) will match any of your device names, because then Asterisk
; might match the wrong device.
; Note: The parameter “username” is not the username and in most cases is
; not needed at all. Check below. In later releases, it’s renamed
; to “defaultuser” which is a better name, since it is used in
; combination with the “defaultip” setting.

; ** Old configuration options **
; The “call-limit” configuation option is considered old is replaced
; by new functionality. To enable callcounters, you use the new
; “callcounter” setting (for extension states in queue and subscriptions)
; You are encouraged to use the dialplan groupcount functionality
; to enforce call limits instead of using this channel-specific method.
; You can still set limits per device in sip.conf or in a database by using
; “setvar” to set variables that can be used in the dialplan for various limits.

context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
; If your Asterisk is connected to the Internet
; and you have allowguest=yes
; you want to check which services you offer everyone
; out there, by enabling them in the default context (see below).
;match_auth_username=yes ; if available, match user entry using the
; ‘username’ field from the authentication line
; instead of the From: field.
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
; Default is enabled. The Dial() options ‘t’ and ‘T’ are not
; related as to whether SIP transfers are allowed or not.
;realm=mydomain.tld ; Realm for digest authentication
; defaults to “asterisk”. If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
;domainsasrealm=no ; Use domans list as realms
; You can serve multiple Realms specifying several
; ‘domain=…’ directives (see below).
; In this case Realm will be based on request ‘From’/‘To’ header
; and should match one of domain names.
; Otherwise default ‘realm=…’ will be used.

; With the current situation, you can do one of four things:
; a) Listen on a specific IPv4 address. Example: bindaddr=
; b) Listen on a specific IPv6 address. Example: bindaddr=2001:db8::1
; c) Listen on the IPv4 wildcard. Example: bindaddr=
; d) Listen on the IPv4 and IPv6 wildcards. Example: bindaddr=::
; (You can choose independently for UDP, TCP, and TLS, by specifying different values for
; “udpbindaddr”, “tcpbindaddr”, and “tlsbindaddr”.)
; (Note that using bindaddr=:: will show only a single IPv6 socket in netstat.
; IPv4 is supported at the same time using IPv4-mapped IPv6 addresses.)
; You may optionally add a port number. (The default is port 5060 for UDP and TCP, 5061
; for TLS).
; IPv4 example: bindaddr=
; IPv6 example: bindaddr=[::]:5062
; The address family of the bound UDP address is used to determine how Asterisk performs
; DNS lookups. In cases a) and c) above, only A records are considered. In case b), only
; AAAA records are considered. In case d), both A and AAAA records are considered. Note,
; however, that Asterisk ignores all records except the first one. In case d), when both A
; and AAAA records are available, either an A or AAAA record will be first, and which one
; depends on the operating system. On systems using glibc, AAAA records are given
; priority.

udpbindaddr= ; IP address to bind UDP listen socket to ( binds to all)
; Optionally add a port number, (default is port 5060)

; When a dialog is started with another SIP endpoint, the other endpoint
; should include an Allow header telling us what SIP methods the endpoint
; implements. However, some endpoints either do not include an Allow header
; or lie about what methods they implement. In the former case, Asterisk
; makes the assumption that the endpoint supports all known SIP methods.
; If you know that your SIP endpoint does not provide support for a specific
; method, then you may provide a comma-separated list of methods that your
; endpoint does not implement in the disallowed_methods option. Note that
; if your endpoint is truthful with its Allow header, then there is no need
; to set this option. This option may be set in the general section or may
; be set per endpoint. If this option is set both in the general section and
; in a peer section, then the peer setting completely overrides the general
; setting (i.e. the result is not the union of the two options).
; Note also that while Asterisk currently will parse an Allow header to learn
; what methods an endpoint supports, the only actual use for this currently
; is for determining if Asterisk may send connected line UPDATE requests. Its
; use may be expanded in the future.
; disallowed_methods = UPDATE

; Note that the TCP and TLS support for chan_sip is currently considered
; experimental. Since it is new, all of the related configuration options are
; subject to change in any release. If they are changed, the changes will
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
tcpenable=no ; Enable server for incoming TCP connections (default is no)
tcpbindaddr= ; IP address for TCP server to bind to ( binds to all interfaces)
; Optionally add a port number, (default is port 5060)

;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr= ; IP address for TLS server to bind to ( binds to all interfaces)
; Optionally add a port number, (default is port 5061)
; Remember that the IP address must match the common name (hostname) in the
; certificate, so you don’t want to bind a TLS socket to multiple IP addresses.
; For details how to construct a certificate for SIP see

;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
; of seconds a client has to authenticate. If
; the client does not authenticate beofre this
; timeout expires, the client will be
; disconnected. (default: 30 seconds)

;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
; unauthenticated sessions that will be allowed
; to connect at any given time. (default: 100)

srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
; Specifying a port in a SIP peer definition or
; when dialing outbound calls will supress SRV
; lookups for that peer or call.

;pedantic=yes ; Enable checking of tags in headers,
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to “yes”)

; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;tos_text=af41 ; Sets TOS for RTP text packets.

;cos_sip=3 ; Sets 802.1p priority for SIP packets.
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
;cos_video=4 ; Sets 802.1p priority for RTP video packets.
;cos_text=3 ; Sets 802.1p priority for RTP text packets.

;maxexpiry=3600 ; Maximum allowed time of incoming registrations
; and subscriptions (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
; Default value is 70
;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
; Set to low value if you use low timeout for NAT of UDP sessions
; Default: 60
;qualifygap=100 ; Number of milliseconds between each group of peers being qualified
; Default: 100
;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
; Default: 1
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;buggymwi=no ; Cisco SIP firmware doesn’t support the MWI RFC
; fully. Enable this option to not get error messages
; when sending MWI to phones with this bug.
;mwi_from=asterisk ; When sending MWI NOTIFY requests, use this setting in
; the From: header as the “name” portion. Also fill the
; “user” portion of the URI in the From: header with this
; value if no fromuser is set
; Default: empty
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to “asterisk”

; Codec negotiation
; When Asterisk is receiving a call, the codec will initially be set to the
; first codec in the allowed codecs defined for the user receiving the call
; that the caller also indicates that it supports. But, after the caller
; starts sending RTP, Asterisk will switch to using whatever codec the caller
; is sending.
; When Asterisk is placing a call, the codec used will be the first codec in
; the allowed codecs that the callee indicates that it supports. Asterisk will
; not switch to whatever codec the callee is sending.
;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec
; rather than advertising all joint codec capabilities. This
; limits the other side’s codec choice to exactly what we prefer.

;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ; see doc/rtp-packetization for framing options
; This option specifies a preference for which music on hold class this channel
; should listen to when put on hold if the music class has not been set on the
; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
; channel putting this one on hold did not suggest a music class.
; This option may be specified globally, or on a per-user or per-peer basis.
; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold. It may be specified globally or on
; a per-user or per-peer basis.
;parkinglot=plaza ; Sets the default parking lot for call parking
; This may also be set for individual users/peers
; Parkinglots are configured in features.conf
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no)
;sendrpid = rpid ; Use the “Remote-Party-ID” header
; to send the identity of the remote party
; This is identical to sendrpid=yes
;sendrpid = pai ; Use the “P-Asserted-Identity” header
; to send the identity of the remote party
;rpid_update = no ; In certain cases, the only method by which a connected line
; change may be immediately transmitted is with a SIP UPDATE request.
; If communicating with another Asterisk server, and you wish to be able
; transmit such UPDATE messages to it, then you must enable this option.
; Otherwise, we will have to wait until we can send a reinvite to
; transmit the information.
;prematuremedia=no ; Some ISDN links send empty media frames before
; the call is in ringing or progress state. The SIP
; channel will then send 183 indicating early media
; which will be empty - thus users get no ring signal.
; Setting this to “yes” will stop any media before we have
; call progress (meaning the SIP channel will not send 183 Session
; Progress for early media). Default is “yes”. Also make sure that
; the SIP peer is configured with progressinband=never.
; In order for “noanswer” applications to work, you need to run
; the progress() application in the priority before the app.

;progressinband=never ; If we should generate in-band ringing always
; use ‘never’ to never use in-band signalling, even in cases
; where some buggy devices might not render it
; Valid values: yes, no, never Default: never
;useragent=Asterisk PBX ; Allows you to change the user agent string
; The default user agent string also contains the Asterisk
; version. If you don’t want to expose this, change the
; useragent string.
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Note that promiscredir when redirects are made to the
; local system will cause loops since Asterisk is incapable
; of performing a “hairpin” call.
;usereqphone = no ; If yes, “;user=phone” is added to uri that contains
; a valid phone number
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:
; info : SIP INFO messages (application/dtmf-relay)
; shortinfo : SIP INFO messages (application/dtmf)
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise

;compactheaders = yes ; send compact sip headers.
;videosupport=yes ; Turn on support for SIP video. You need to turn this
; on in this section to get any video support at all.
; You can turn it off on a per peer basis if the general
; video support is enabled, but you can’t enable it for
; one peer only without enabling in the general section.
; If you set videosupport to “always”, then RTP ports will
; always be set up for video, even on clients that don’t
; support it. This assists callfile-derived calls and
; certain transferred calls to use always use video when
; available. [yes|NO|always]

;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
; Videosupport and maxcallbitrate is settable
; for peers and users as well
;callevents=no ; generate manager events when sip ua
; performs events (e.g. hold)
;authfailureevents=no ; generate manager “peerstatus” events when peer can’t
; authenticate with Asterisk. Peerstatus will be “rejected”.
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
; for any reason, always reject with an identical response
; equivalent to valid username and invalid password/hash
; instead of letting the requester know whether there was
; a matching user or peer for their request. This reduces
; the ability of an attacker to scan for valid SIP usernames.
; This option is set to “yes” by default.

;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
; INVITE requests are. By default this option is disabled.

;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
; order instead of RFC3551 packing order (this is required
; for Sipura and Grandstream ATAs, among others). This is
; contrary to the RFC3551 specification, the peer should
; be negotiating AAL2-G726-32 instead :frowning:
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
;outboundproxy=tls://proxy.provider.domain ; same as ‘=proxy.provider.domain’ except we try to connect with tls
;outboundproxy= ; IPv4 address literal (default port is 5060)
;outboundproxy=2001:db8::1 ; IPv6 address literal (default port is 5060)
;outboundproxy= ; IPv4 address literal with explicit port
;outboundproxy=[2001:db8::1]:5062 ; IPv6 address literal with explicit port
; ; (could also be tcp,udp) - defining transports on the proxy line only
; ; applies for the global proxy, otherwise use the transport= option
;matchexternaddrlocally = yes ; Only substitute the externaddr or externhost setting if it matches
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.

;dynamic_exclude_static = yes ; Disallow all dynamic hosts from registering
; as any IP address used for staticly defined
; hosts. This helps avoid the configuration
; error of allowing your users to register at
; the same address as a SIP provider.

;contactdeny= ; Use contactpermit and contactdeny to
;contactpermit= ; restrict at what IPs your users may
; register their phones.

;engine=asterisk ; RTP engine to use when communicating with the device

; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with
; us and have a “regexten=” configuration item.
; Multiple contexts may be specified by separating them with ‘&’. The
; actual extension is the ‘regexten’ parameter of the registering peer or its
; name if ‘regexten’ is not provided. If more than one context is provided,
; the context must be specified within regexten by appending the desired
; context after ‘@’. More than one regexten may be supplied if they are
; separated by ‘&’. Patterns may be used in regexten.
;regextenonqualify=yes ; Default “no”
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer

;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite media streams to an optimal path. If there’s
; no reason for Asterisk to stay in the media path, the media will be redirected.
; This does not really work well in the case where Asterisk is outside and the
; clients are on the inside of a NAT. In that case, you want to set directmedia=nonat.
;directmedia=yes ; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason want Asterisk to
; stay in the audio path, you may want to turn this off.

                            ; This setting also affect direct RTP
                            ; at call setup (a new feature in 1.4 - setting up the
                            ; call directly between the endpoints instead of sending
                            ; a re-INVITE).

                            ; Additionally this option does not disable all reINVITE operations.
                            ; It only controls Asterisk generating reINVITEs for the specific
                            ; purpose of setting up a direct media path. If a reINVITE is
                            ; needed to switch a media stream to inactive (when placed on
                            ; hold) or to T.38, it will still be done, regardless of this 
                            ; setting. Note that direct T.38 is not supported.

;directmedia=nonat ; An additional option is to allow media path redirection
; (reinvite) but only when the peer where the media is being
; sent is known to not be behind a NAT (as the RTP core can
; determine it based on the apparent IP address the media
; arrives from).

;directmedia=update ; Yet a third option… use UPDATE for media path redirection,
; instead of INVITE. This can be combined with ‘nonat’, as
; ‘directmedia=update,nonat’. It implies ‘yes’.

;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
; the call directly with media peer-2-peer without re-invites.
; Will not work for video and cases where the callee sends
; RTP payloads and fmtp headers in the 200 OK that does not match the
; callers INVITE. This will also fail if directmedia is enabled when
; the device is actually behind NAT.

;directmediadeny= ; Use directmediapermit and directmediadeny to restrict
;directmediapermit=; which peers should be able to pass directmedia to each other
; (There is no default setting, this is just an example)
; Use this if some of your phones are on IP addresses that
; can not reach each other directly. This way you can force
; RTP to always flow through asterisk in such cases.

;ignoresdpversion=yes ; By default, Asterisk will honor the session version
; number in SDP packets and will only modify the SDP
; session if the version number changes. This option will
; force asterisk to ignore the SDP session version number
; and treat all SDP data as new data. This is required
; for devices that send us non standard SDP packets
; (observed with Microsoft OCS). By default this option is
; off.

;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
; Like the useragent parameter, the default user agent string
; also contains the Asterisk version.
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
; This field MUST NOT contain spaces
;encryption=no ; Whether to offer SRTP encrypted media (and only SRTP encrypted media)
; on outgoing calls to a peer. Calls will fail with HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.

; The SIP channel has two types of devices, the friend and the peer.
; * The type=friend is a device type that accepts both incoming and outbound calls,
; where Asterisk match on the From: username on incoming calls.
; (A synonym for friend is “user”). This is a type you use for your local
; SIP phones.
; * The type=peer also handles both incoming and outbound calls. On inbound calls,
; Asterisk only matches on IP/port, not on names. This is mostly used for SIP
; trunks.
; For device names, we recommend using only a-z, numerics (0-9) and underscore
; For local phones, type=friend works most of the time
; If you have one-way audio, you probably have NAT problems.
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
; Configuration options available
; --------------------
; context
; callingpres
; permit
; deny
; secret
; md5secret
; remotesecret
; transport
; dtmfmode
; directmedia
; nat
; callgroup
; pickupgroup
; language
; allow
; disallow
; insecure
; trustrpid
; progressinband
; promiscredir
; useclientcode
; accountcode
; setvar
; callerid
; amaflags
; callcounter
; busylevel
; allowoverlap
; allowsubscribe
; allowtransfer
; ignoresdpversion
; subscribecontext
; template
; videosupport
; maxcallbitrate
; rfc2833compensate
; mailbox
; session-timers
; session-expires
; session-minse
; session-refresher
; t38pt_usertpsource
; regexten
; fromdomain
; fromuser
; host
; port
; qualify
; defaultip
; defaultuser
; rtptimeout
; rtpholdtimeout
; sendrpid
; outboundproxy
; rfc2833compensate
; callbackextension
; registertrying
; timert1
; timerb
; qualifyfreq
; t38pt_usertpsource
; contactpermit ; Limit what a host may register as (a neat trick
; contactdeny ; is to register at the same IP as a SIP provider,
; ; then call oneself, and get redirected to that
; ; same location).
; directmediapermit
; directmediadeny
; unsolicited_mailbox
; use_q850_reason
; maxforwards
; encryption

; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)

;type=peer ; we only want to call out, not be called
;remotesecret=guessit ; Our password to their service
;defaultuser=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;transport=udp,tcp ; This sets the default transport type to udp for outgoing, and will
; ; accept both tcp and udp. The default transport type is only used for
; ; outbound messages until a Registration takes place. During the
; ; peer Registration the transport type may change to another supported
; ; type if the peer requests so.

;usereqphone=yes ; This provider requires “;user=phone” on URI
;callcounter=yes ; Enable call counter
;busylevel=2 ; Signal busy at 2 or more calls
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
;port=80 ; The port number we want to connect to on the remote side
; Also used as “defaultport” in combination with “defaultip” settings

;— sample definition for a provider
;fromuser=4015552299 ; how your provider knows you
;remotesecret=youwillneverguessit ; The password we use to authenticate to them
;secret=gissadetdu ; The password they use to contact us
;callbackextension=123 ; Register with this server and require calls coming back to this extension
;transport=udp,tcp ; This sets the transport type to udp for outgoing, and will
; ; accept both tcp and udp. Default is udp. The first transport
; ; listed will always be used for outgoing connections.
;unsolicited_mailbox=4015552299 ; If the remote SIP server sends an unsolicited MWI NOTIFY message the new/old
; ; message count will be stored in the configured virtual mailbox. It can be used
; ; by any device supporting MWI by specifying @SIP_Remote as the
; ; mailbox.

; Because you might have a large number of similar sections, it is generally
; convenient to use templates for the common parameters, and add them
; the the various sections. Examples are below, and we can even leave
; the templates uncommented as they will not harm:

basic-options ; a template

natted-phone ; another template inheriting basic-options

public-phone ; another template inheriting basic-options

my-codecs ; a template for my preferred codecs

ulaw-phone ; and another one for ulaw-only

; and finally instantiate a few phones
; 2133
; secret = peekaboo
; 2134
; secret = not_very_secret
; 2136
; secret = not_very_secret_either
; …

; Standard configurations not using templates look like this:
;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
; on incoming calls to Asterisk
;host= ; we have a static but private IP address
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
;directmedia=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
; from the phone to asterisk (deprecated)
; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory
; There is no combined call counter for a “friend”
; so there’s currently no way in sip.conf to limit
; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.
;mailbox=1234@default ; mailbox 1234 in voicemail context “default”
;disallow=all ; need to disallow=all before we can use allow=
;allow=ulaw ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
; See README.callingpres for more information

; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;regexten=1234 ; When they register, create extension 1234
;callerid=“Jane Smith” <5678>
;host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
;directmedia=no ; Typically set to NO if behind NAT
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
;registertrying=yes ; Send a 100 Trying when the device registers.

;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
;language=de ; Use German prompts for this user
;host=dynamic ; This peer register with us
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;defaultip= ; IP used until peer registers
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
;subscribemwi=yes ; Only send notifications if this phone
; subscribes for mailbox notification
;vmexten=voicemail ; dialplan extension to reach mailbox
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to “asterisk”
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!

;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;host=dynamic ; This peer register with us
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;defaultuser=polly ; Username to use in INVITE until peer registers
; Normally you do NOT need to set this parameter
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no ; Polycom phones don’t work properly with “never”

;insecure=port ; Allow matching of peer by IP address without
; matching port number
;insecure=invite ; Do not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
;qualify=1000 ; Consider it down if it’s 1 second to reply
; Helps with NAT session
; qualify=yes uses default value
;qualifyfreq=60 ; Qualification: How often to check for the
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
; Call group and Pickup group should be in the range from 0 to 63
;callgroup=1,3-4 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
;defaultip= ; IP address to use if peer has not registered
;deny= ; ACL: Control access to this account based on IP address
;permit= ; we can also use CIDR notation for subnet masks
;permit=2001:db8::/32 ; IPv6 ACLs can be specified if desired. IPv6 ACLs
; apply only to IPv6 addresses, and IPv4 ACLs apply
; only to IPv4 addresses.




;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
; You must have this turned on or DTMF reception will work improperly.
;t38pt_usertpsource=yes ; Use the source IP address of RTP as the destination IP address for UDPTL packets
; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
; external IP address of the remote device. If port forwarding is done at the client side
; then UDPTL will flow to the remote device.[/code]

8000 is Cisco SPA504G

8001 and 8002 Softphones, there work fine, but Cisco 8000 not register, this is output of sip debug:

[code]Reloading SIP
Reliably Transmitting (no NAT) to
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK6180acc4
Max-Forwards: 70
From: “asterisk” sip:asterisk@;tag=as7e90a2b2
To: sip:
Contact: sip:asterisk@
Call-ID: 0c118f41690916997250baef600a3ec7@
User-Agent: Asterisk PBX
Date: Thu, 05 May 2011 01:42:35 GMT
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP: —>
OPTIONS sip: SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK6180acc4
Max-Forwards: 70
From: “asterisk” sip:asterisk@;tag=as7e90a2b2
To: sip:
Contact: sip:asterisk@
Call-ID: 0c118f41690916997250baef600a3ec7@
User-Agent: Asterisk PBX
Date: Thu, 05 May 2011 01:42:35 GMT
Supported: replaces, timer
Content-Length: 0

— (13 headers 0 lines) —
Looking for in default (domain

<— Transmitting (no NAT) to —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP;branch=z9hG4bK6180acc4;received=
From: “asterisk” sip:asterisk@;tag=as7e90a2b2
To: sip:;tag=as03233196
Call-ID: 0c118f41690916997250baef600a3ec7@
Server: Asterisk PBX
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

Scheduling destruction of SIP dialog ‘0c118f41690916997250baef600a3ec7@’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP: —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP;branch=z9hG4bK6180acc4;received=
From: “asterisk” sip:asterisk@;tag=as7e90a2b2
To: sip:;tag=as03233196
Call-ID: 0c118f41690916997250baef600a3ec7@
Server: Asterisk PBX
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

— (11 headers 0 lines) —
Really destroying SIP dialog ‘0c118f41690916997250baef600a3ec7@’ Method: OPTIONS