Re: Ayuda con Asterisk
2011/5/20 Douglas Flores <douglasfloresnic@gmail.com>:
> Sería bueno ver los archivos de configuración para saber que fue lo que
> hicistes.
>
> saludos.
>
>
>
> El 20 de mayo de 2011 12:10, Maximiliano Marin Bustos <maxmarin@gmail.com>
> escribió:
>>
>> Hola Gente! Tengo el siguiente problema con Asterisk.
>> Instale asterisk via apt y cree algunos anexos y usuarios en sip.conf
>> y extensions.conf.
>> El punto es que cuando me quiero conectar por zoiper (softphone), no
>> se conecta nunca y sale "registering" y de ahi no sale.
>> Asterisk me dice esto:
>>
>> Aristoteles*CLI> sip show users
>> Username Secret Accountcode Def.Context ACL NAT
>> 81 81 internal No RFC3581
>> 80 80 internal No RFC3581
>> Aristoteles*CLI>
>>
>> No se que hacer.
>> Quien me echa una manito?
>>
>>
>> --
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>> Archive:
>> [🔎] BANLkTimNXsP2ovjOLnf2=0PEUfQ3+yAXxg@mail.gmail.com">http://lists.debian.org/[🔎] BANLkTimNXsP2ovjOLnf2=0PEUfQ3+yAXxg@mail.gmail.com
>>
>
>
>
> --
> Ing. Douglas Flores.
> "Linux no es Alternativa, es Solución."
> Asterisk Users #1009.
>
Hola Douglas:
Gracias por su respuesta. Los archivos de configuracion son los siguientes:
; 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.
;
[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 "dialplan save" too
;
writeprotect=no
;
; 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).
;
;autofallthrough=no
;
;
;
; 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 bugs.digium.com. 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.
;
;extenpatternmatchnew=no
;
; 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.
;
clearglobalvars=no
;
; 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.
;
;priorityjumping=yes
;
; User context is where entries from users.conf are registered. The
; default value is 'default'
;
;userscontext=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/build-extra-contexts.sh
;#exec /opt/bin/build-extra-contexts.sh --foo="bar"
;#exec </opt/bin/build-extra-contexts.sh --foo="bar">
;#exec "/opt/bin/build-extra-contexts.sh --foo=\"bar\""
;
; The "Globals" category contains global variables that can be referenced
; in the dialplan with the GLOBAL dialplan function:
; ${GLOBAL(VARIABLE)}
; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
; Unix/Linux environmental variables can be reached with the ENV dialplan
; function: ${ENV(VARIABLE)}
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
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)
;TRUNK=IAX2/user:pass@provider
;FREENUMDOMAIN=mydomain.com ; domain to send on outbound
; freenum calls (uses
outbound-freenum
; context)
;
; WARNING WARNING WARNING WARNING
; If you load any other extension configuration engine, such as pbx_ael.so,
; 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.
; WARNING WARNING WARNING WARNING
;
; 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".
;
;[context]
;exten => someexten,{priority|label{+|-}offset}[(alias)],application(arg1,arg2,...)
;
; Timing list for includes is
;
; <time range>,<days of week>,<days of month>,<months>[,<timezone>]
;
; 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
;
;
[dundi-e164-canonical]
;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})
[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)
[dundi-e164-via-pstn]
;
; 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
[dundi-e164-local]
;
; 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
[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164
[dundi-e164-lookup]
;
; 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.
;
[macro-dundi-e164]
;
; 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(${ARG1},1)
include => dundi-e164-lookup
;
; Here are the entries you need to participate in the IAXTEL
; call routing system. Most IAXTEL numbers begin with 1-700, but
; there are exceptions. For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel)
;
; The SWITCH statement permits a server to share the dialplan with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext
[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})})
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${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:${GLOBAL(TRUNKMSD)}})
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
;Include parkedcalls (or the context you define in features conf)
;to enable call parking.
include => parkedcalls
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password@bigserver/local
;
; An "lswitch" is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
;
; lswitch => Loopback/12${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@${CURSERVER}
; The following two contexts are a template to enable the ability to dial
; ISN numbers. For more information about what an ISN number is, please see
; http://www.freenum.org.
;
; This is the dialing hook. use:
; include => outbound-freenum
[outbound-freenum]
; We'll add more digits as needed. The purpose is to dial things
; like extension numbers at domains (ITAD number) so we're matching
; on lengths of 1 through 6 prior to the separator (the asterisk [*])
;
exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
[outbound-freenum2]
; 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
README-SERIOUSLY.best-practices.txt document
same => n,Set(TIMEOUT(absolute)=10800)
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
; perform our lookup with freenum.org
same => n,GotoIf($["${isnresult}" != ""]?from)
same => n,Set(DIALSTATUS=CONGESTION)
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()
[macro-trunkdial]
;
; 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
[stdexten]
;
; Standard extension subroutine:
; ${EXTEN} - Extension
; ${ARG1} - Device(s) to ring
; ${ARG2} - Optional context in Voicemail (if empty, then "default")
;
; Note that the current version will drop through to the next priority in the
; case of their pressing '#'. This gives more flexibility in what do to next:
; you can prompt for a new extension, or drop the call, or send them to a
; general delivery mailbox, or...
;
; The use of the LOCAL() function is purely for convenience. Any variable
; initially declared as LOCAL() will disappear when the innermost Gosub context
; in which it was declared returns. Note also that you can declare a LOCAL()
; variable on top of an existing variable, and its value will revert to its
; previous value (before being declared as LOCAL()) upon Return.
;
exten => _X.,50000(stdexten),NoOp(Start stdexten)
exten => _X.,n,Set(LOCAL(ext)=${EXTEN})
exten => _X.,n,Set(LOCAL(dev)=${ARG1})
exten => _X.,n,Set(LOCAL(cntx)=${ARG2})
exten => _X.,n,Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${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(${mbx},u) ; If unavailable,
send to voicemail w/ unavail announce
exten => stdexten-NOANSWER,n,NoOp(Finish stdexten NOANSWER)
exten => stdexten-NOANSWER,n,Return() ; If they press #, return to start
exten => stdexten-BUSY,1,Voicemail(${mbx},b)
; If busy, send to voicemail w/ busy announce
exten => stdexten-BUSY,n,NoOp(Finish stdexten BUSY)
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()
[stdPrivacyexten]
;
; 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
[macro-page];
;
; 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([${AVAILORIGCHAN} = ""]?fail:autoanswer)
exten => s,n(autoanswer),Set(_ALERT_INFO="RA") ; This is for the PolyComs
exten => s,n,SIPAddHeader(Call-Info: Answer-After=0) ; This is for the
Grandstream, Snoms, and Others
exten => s,n,NoOp() ; Add others here and Post on the Wiki!!!!
exten => s,n,Dial(${ARG1})
exten => s,n(fail),Hangup
[demo]
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/guest@pbx.digium.com/s@default) ; 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/${EXTEN})
;
;
; System Wide Page at extension 7999
;
exten => 7999,1,Set(TIMEOUT(absolute)=60)
exten => 7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d)
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)
;
; The page context calls up the page macro that sets variables needed
for auto-answer
; It is in is own context to make calling it from the Page()
application as simple as
; Local/{peername}@page
;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})
;[mainmenu]
;
; 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
;
;[submenu]
;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)
[default]
;
; 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@192.168.0.14)
;exten => 6390,1,Dial(JINGLE/caller/callee) ; Dial via jingle using labels
;exten => 6391,1,Dial(JINGLE/asterisk@digium.com/mogorman@astjab.org)
;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...
;[acme-incoming]
;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)
;
;[acme-internal]
;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.
;
;[acme-extens]
;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.
;
[time]
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
;
[ani]
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 <command>" 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 <COMMAND>" will show you more information about
; one function. Remember that function names are UPPER CASE.
[internal]
;
exten => 80,1,Dial(SIP/80,26)
exten => 80,n,Hangup
;
exten => 81,1,Dial(SIP/81,26)
exten => 81,n,Hangup
;
sip.conf
;
; SIP Configuration example for Asterisk
;
; 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
;
;
; 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[:authname]]]@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/sales:topsecret::account02@domain.com:5062
; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@192.168.0.1
;
; 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!sales@edvina.net
;
; 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
;
; module reload chan_sip.so Reload configuration file
;
;------- 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.
;-----------------------------------------------------------------------------
; ** Deprecated configuration options **
; The "call-limit" configuation option is deprecated. It still works in
; this version of Asterisk, but will disappear in the next version.
; 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.
[general]
;
port=5060
disallow=all
allow=g726
allow=ulaw
allow=alaw
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
;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
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket
to (0.0.0.0 binds to all)
; Optionally add a port number,
192.168.1.1:5062 (default is port 5060)
;
; 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=0.0.0.0 ; IP address for TCP server to bind to
(0.0.0.0 binds to all interfaces)
; Optionally add a port number,
192.168.1.1:5062 (default is port 5060)
;tlsenable=no ; Enable server for incoming TLS
(secure) connections (default is no)
;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to
(0.0.0.0) binds to all interfaces)
; Optionally add a port number,
192.168.1.1:5063 (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
;
http://tools.ietf.org/html/draft-ietf-sip-domain-certs
;tlscertfile=asterisk.pem ; Certificate file (*.pem only) to use
for TLS connections
; default is to look for
"asterisk.pem" in current directory
;tlscafile=</path/to/certificate>
; If the server your connecting to uses a self signed certificate
; you should have their certificate installed here so the code can
; verify the authenticity of their certificate.
;tlscadir=</path/to/ca/dir>
; A directory full of CA certificates. The files must be named with
; the CA subject name hash value.
; (see man SSL_CTX_load_verify_locations for more info)
;tlsdontverifyserver=[yes|no]
; If set to yes, don't verify the servers certificate when acting as
; a client. If you don't have the server's CA certificate you can
; set this and it will connect without requiring tlscafile to be set.
; Default is no.
;tlscipher=<SSL cipher string>
; A string specifying which SSL ciphers to use or not use
; A list of valid SSL cipher strings can be found at:
; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS
;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 "no")
; 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
;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
;qualifygap=100 ; Number of milliseconds between each group of peers
being qualified
;qualifypeers=1 ; Number of peers in a group to be qualified at the same time
;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.
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"
;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.
;
;mohinterpret=default
;
; 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.
;
;mohsuggest=default
;
;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
;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 "no" will stop any media before we have
; call progress. Default is "yes".
;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.
;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
;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.
;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 :-(
;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
; ; (could also be
tcp,udp) - defining transports on the proxy line only
; ; applies for the
global proxy, otherwise use the transport= option
;matchexterniplocally = yes ; Only substitute the externip 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=0.0.0.0/0.0.0.0 ; Use contactpermit and contactdeny to
;contactpermit=172.16.0.0/255.255.0.0 ; restrict at what IPs your users may
; register their phones.
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.',
and '-' not
; in square brackets. For example, the caller id value 555.5555 becomes 5555555
; when this option is enabled. Disabling this option results in no modification
; of the caller id value, which is necessary when the caller id
represents something
; that must be preserved. This option can only be used in the
[general] section.
; By default this option is on.
;
;shrinkcallerid=yes ; on by default
;
; 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.
;
;regcontext=sipregistrations
;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
;
;--------------------------- SIP timers
----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
;
;t1min=100 ; Minimum roundtrip time for messages
to monitored hosts
; Defaults to 100 ms
;timert1=500 ; Default T1 timer
; Defaults to 500 ms or the measured round-trip
; time to a peer (qualify=yes).
;timerb=32000 ; Call setup timer. If a provisional
response is not received
; in this amount of time, the call
will autocongest
; Defaults to 64*timert1
;--------------------------- RTP timers
----------------------------------------------------
; These timers are currently used for both audio and video streams.
The RTP timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;
;rtptimeout=60 ; Terminate call if 60 seconds of no
RTP or RTCP activity
; on the audio channel
; when we're not on hold. This is to
be able to hangup
; a call in the case of a phone
disappearing from the net,
; like a powerloss or grandma tripping
over a cable.
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no
RTP or RTCP activity
; on the audio channel
; when we're on hold (must be > rtptimeout)
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to
keep NAT open
; (default is off - zero)
;--------------------------- SIP Session-Timers (RFC
4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for
active SIP sessions.
; This mechanism can detect and reclaim SIP channels that do not
terminate through normal
; signaling procedures. Session-Timers can be configured globally or
at a user/peer level.
; The operation of Session-Timers is driven by the following
configuration parameters:
;
; * session-timers - Session-Timers feature operates in the
following three modes:
; originate : Request and run session-timers always
; accept : Run session-timers only when
requested by other UA
; refuse : Do not run session timers in any case
; The default mode of operation is 'accept'.
; * session-expires - Maximum session refresh interval in seconds.
Defaults to 1800 secs.
; * session-minse - Minimum session refresh interval in seconds.
Defualts to 90 secs.
; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
;
;session-timers=originate
;session-expires=600
;session-minse=90
;session-refresher=uas
;
;--------------------------- HASH TABLE SIZES
------------------------------------------------
; For maximum efficiency, adjust the following
; values to be slightly larger than the maximum number of in-memory
objects (devices).
; Too large, and space is wasted. Too small, and things will run slower.
; 563 is probably way too big for small (home) applications, but it
; should cover most small/medium sites.
; It is recommended to make the sizes be a prime number!
; This was internally set to 17 for small-memory applications...
; All tables default to 563, except when compiled in LOW_MEMORY mode,
; in which case, they default to 17. You can override this by uncommenting
; the following, and changing the values.
;hash_users=563
;hash_peers=563
;hash_dialogs=563
;--------------------------- SIP DEBUGGING
---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this
configuration
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
; SIP history is output to the DEBUG
logging channel
;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS)
----------------------------
; You can subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE
;
; You will get more detailed reports (busy etc) if you have a call
counter enabled
; for a device.
;
; If you set the busylevel, we will indicate busy when we have a
number of calls that
; matches the busylevel treshold.
;
; For queues, you will need this level of detail in status reporting, regardless
; if you use SIP subscriptions. Queues and manager use the same
internal interface
; for reading status information.
;
; Note: Subscriptions does not work if you have a realtime dialplan and use the
; realtime switch.
;
;allowsubscribe=no ; Disable support for subscriptions.
(Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to
local extensions
; Settable per peer/user also
;notifyringing = no ; Control whether subscriptions
already INUSE get sent
; RINGING when another call is sent
(default: yes)
;notifyhold = yes ; Notify subscriptions on HOLD state
(default: no)
; Turning on notifyringing and
notifyhold will add a lot
; more database transactions if you
are using realtime.
;notifycid = yes ; Control whether caller ID
information is sent along with
; dialog-info+xml notifications
(supported by snom phones).
; Note that this feature will only
work properly when the
; incoming call is using the same
extension and context that
; is being used as the hint for the
called extension. This means
; that it won't work when using
subscribecontext for your sip
; user or peer (if subscribecontext is
different than context).
; This is also limited to a single
caller, meaning that if an
; extension is ringing because
multiple calls are incoming,
; only one will be used as the source
of caller ID. Specify
; 'ignore-context' to ignore the
called context when looking
; for the caller's channel. The
default value is 'no.' Setting
; notifycid to 'ignore-context' also
causes call-pickups attempted
; via SNOM's NOTIFY mechanism to set
the context for the call pickup
; to PICKUPMARK.
;callcounter = yes ; Enable call counters on devices.
This can be set per
; device too.
;----------------------------------------- T.38 FAX SUPPORT
----------------------------------
;
; This setting is available in the [general] section as well as in
device configurations.
; Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off.
;
; t38pt_udptl = yes ; Enables T.38 with FEC error correction.
; t38pt_udptl = yes,fec ; Enables T.38 with FEC error correction.
; t38pt_udptl = yes,redundancy ; Enables T.38 with redundancy error correction.
; t38pt_udptl = yes,none ; Enables T.38 with no error correction.
;
; In some cases, T.38 endpoints will provide a T38FaxMaxDatagram value
(during T.38 setup) that
; is based on an incorrect interpretation of the T.38 recommendation,
and results in failures
; because Asterisk does not believe it can send T.38 packets of a
reasonable size to that
; endpoint (Cisco media gateways are one example of this situation).
In these cases, during a
; T.38 call you will see warning messages on the console/in the logs
from the Asterisk UDPTL
; stack complaining about lack of buffer space to send T.38 FAX
packets. If this occurs, you
; can set an override (globally, or on a per-device basis) to make
Asterisk ignore the
; T38FaxMaxDatagram value specified by the other endpoint, and use a
configured value instead.
; This can be done by appending 'maxdatagram=<value>' to the
t38pt_udptl configuration option,
; like this:
;
; t38pt_udptl = yes,fec,maxdatagram=400 ; Enables T.38 with FEC error
correction and overrides
; ; the other endpoint's
provided value to assume we can
; ; send 400 byte T.38 FAX packets to it.
;
; FAX detection will cause the SIP channel to jump to the 'fax'
extension (if it exists)
; based one or more events being detected. The events that can be
detected are an incoming
; CNG tone or an incoming T.38 re-INVITE request.
;
; faxdetect = yes ; Default 'no', 'yes' enables both CNG and T.38 detection
; faxdetect = cng ; Enables only CNG detection
; faxdetect = t38 ; Enables only T.38 detection
; faxdetect = both ; Enables both CNG and T.38 detection (same as 'yes')
;
;----------------------------------------- OUTBOUND SIP REGISTRATIONS
------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register =>
[peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
;
;
;
; domain is either
; - domain in DNS
; - host name in DNS
; - the name of a peer defined below or in realtime
; The domain is where you register your username, so your SIP uri you
are registering to
; is username@domain
;
; If no extension is given, the 's' extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).
;
; A similar effect can be achieved by adding a "callbackextension"
option in a peer section.
; this is equivalent to having the following line in the general section:
;
; register => username:secret@host/callbackextension
;
; and more readable because you don't have to write the parameters in two places
; (note that the "port" is ignored - this is a bug that should be fixed).
;
; Note that a register= line doesn't mean that we will match the
incoming call in any
; other way than described above. If you want to control where the
call enters your
; dialplan, which context, you want to define a peer with the hostname
of the provider's
; server. If the provider has multiple servers to place calls to your
system, you need
; a peer for each server.
;
; Beginning with Asterisk version 1.6.2, the "user" portion of the
register line may
; contain a port number. Since the logical separator between a host
and port number is a
; ':' character, and this character is already used to separate
between the optional "secret"
; and "authuser" portions of the line, there is a bit of a hoop to
jump through if you wish
; to use a port here. That is, you must explicitly provide a "secret"
and "authuser" even if
; they are blank. See the third example below for an illustration.
;
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
; connect to local extension 1234 in extensions.conf, default context,
; unless you configure a [sip_proxy] section below, and configure a
; context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate inbound and outbound sections for SIP providers
; (instead of type=friend) if you have calls in both directions
;
;register => 3456@mydomain:5082::@mysipprovider.com
;
; Note that in this example, the optional authuser and secret portions have
; been left blank because we have specified a port in the user section
;
;register => tls://username:xxxxxx@sip-tls-proxy.example.org
;
; The 'transport' part defaults to 'udp' but may also be 'tcp' or 'tls'.
; Using 'udp://' explicitly is also useful in case the username part
; contains a '/' ('user/name').
;registertimeout=20 ; retry registration calls every 20
seconds (default)
;registerattempts=10 ; Number of registration attempts
before we give up
; 0 = continue forever, hammering the
other server
; until it accepts the registration
; Default is 0 tries, continue forever
;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS
-------------------------
; Asterisk can subscribe to receive the MWI from another SIP server
and store it locally for retrieval
; by other phones.
; Format for the mwi register statement is:
; mwi => user[:secret[:authuser]]@host[:port][/mailbox]
;
; Examples:
;mwi => 1234:password@mysipprovider.com/1234
;
; MWI received will be stored in the 1234 mailbox of the SIP_Remote
context. It can be used by other phones by following the below:
; mailbox=1234@SIP_Remote
;----------------------------------------- NAT SUPPORT ------------------------
;
; WARNING: SIP operation behind a NAT is tricky and you really need
; to read and understand well the following section.
;
; When Asterisk is behind a NAT device, the "local" address (and port) that
; a socket is bound to has different values when seen from the inside or
; from the outside of the NATted network. Unfortunately this address must
; be communicated to the outside (e.g. in SIP and SDP messages), and in
; order to determine the correct value Asterisk needs to know:
;
; + whether it is talking to someone "inside" or "outside" of the
NATted network.
; This is configured by assigning the "localnet" parameter with a list
; of network addresses that are considered "inside" of the NATted network.
; IF LOCALNET IS NOT SET, THE EXTERNAL ADDRESS WILL NOT BE SET CORRECTLY.
; Multiple entries are allowed, e.g. a reasonable set is the following:
;
; localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
; localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
; localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
; localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
;
; + the "externally visible" address and port number to be used when talking
; to a host outside the NAT. This information is derived by one of the
; following (mutually exclusive) config file parameters:
;
; a. "externip = hostname[:port]" specifies a static address[:port] to
; be used in SIP and SDP messages.
; The hostname is looked up only once, when [re]loading sip.conf .
; If a port number is not present, use the "bindport" value (which is
; not guaranteed to work correctly, because a NAT box might remap the
; port number as well as the address).
; This approach can be useful if you have a NAT device where you can
; configure the mapping statically. Examples:
;
; externip = 12.34.56.78 ; use this address.
; externip = 12.34.56.78:9900 ; use this address and port.
; externip = mynat.my.org:12600 ; Public address of my nat box.
;
; b. "externhost = hostname[:port]" is similar to "externip" except
; that the hostname is looked up every "externrefresh" seconds
; (default 10s). This can be useful when your NAT device lets you choose
; the port mapping, but the IP address is dynamic.
; Beware, you might suffer from service disruption when the name server
; resolution fails. Examples:
;
; externhost=foo.dyndns.net ; refreshed periodically
; externrefresh=180 ; change the refresh interval
;
; c. "stunaddr = stun.server[:port]" queries the STUN server specified
; as an argument to obtain the external address/port.
; Queries are also sent periodically every "externrefresh" seconds
; (as a side effect, sending the query also acts as a keepalive for
; the state entry on the nat box):
;
; stunaddr = foo.stun.com:3478
; externrefresh = 15
;
; Note that at the moment all these mechanism work only for the SIP socket.
; The IP address discovered with externip/externhost/STUN is reused for
; media sessions as well, but the port numbers are not remapped so you
; may still experience problems.
;
; NOTE 1: in some cases, NAT boxes will use different port numbers in
; the internal<->external mapping. In these cases, the "externip" and
; "externhost" might not help you configure addresses properly, and you
; really need to use STUN.
;
; NOTE 2: when using "externip" or "externhost", the address part is
; also used as the external address for media sessions.
; If you use "stunaddr", STUN queries will be sent to the same server
; also from media sockets, and this should permit a correct mapping of
; the port numbers as well.
;
; In addition to the above, Asterisk has an additional "nat" parameter to
; address NAT-related issues in incoming SIP or media sessions.
; In particular, depending on the 'nat= ' settings described below, Asterisk
; may override the address/port information specified in the SIP/SDP messages,
; and use the information (sender address) supplied by the network
stack instead.
; However, this is only useful if the external traffic can reach us.
; The following settings are allowed (both globally and in individual sections):
;
; nat = no ; default. Use NAT mode only
according to RFC3581 (;rport)
; nat = yes ; Always ignore info and assume NAT
; nat = never ; Never attempt NAT mode or RFC3581 support
; nat = route ; route = Assume NAT, don't send rport
; ; (work around more UNIDEN bugs)
;----------------------------------- 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).
;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.
; 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'.
;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.
;----------------------------------------- REALTIME SUPPORT
------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the
; source code.
;
;rtcachefriends=yes ; Cache realtime friends by adding
them to the internal list
; just like friends added from the
config file only on a
; as-needed basis? (yes|no)
;rtsavesysname=yes ; Save systemname in realtime database
at registration
; Default= no
;rtupdate=yes ; Send registry updates to database
using realtime? (yes|no)
; If set to yes, when a SIP UA
registers successfully, the ip address,
; the origination port, the
registration period, and the username of
; the UA will be set to database via realtime.
; If not present, defaults to 'yes'.
Note: realtime peers will
; probably not function across reloads
in the way that you expect, if
; you turn this option off.
;rtautoclear=yes ; Auto-Expire friends created on the
fly on the same schedule
; as if it had just registered?
(yes|no|<seconds>)
; If set to yes, when the registration
expires, the friend will
; vanish from the configuration until
requested again. If set
; to an integer, friends expire within
this number of seconds
; instead of the registration interval.
;ignoreregexpire=yes ; Enabling this setting has two functions:
;
; For non-realtime peers, when their
registration expires, the
; information will _not_ be removed
from memory or the Asterisk database
; if you attempt to place a call to
the peer, the existing information
; will be used in spite of it having expired
;
; For realtime peers, when the peer is
retrieved from realtime storage,
; the registration information will be
used regardless of whether
; it has expired or not; if it expires
while the realtime peer
; is still in memory (due to caching
or other reasons), the
; information will not be removed from
realtime storage
;----------------------------------------- SIP DOMAIN SUPPORT
------------------------
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
; domains, each of which can direct the call to a specific context if desired.
; By default, all domains are accepted and sent to the default context or the
; context associated with the user/peer placing the call.
; REGISTER to non-local domains will be automatically denied if a domain
; list is configured.
;
; Domains can be specified using:
; domain=<domain>[,<context>]
; Examples:
; domain=myasterisk.dom
; domain=customer.com,customer-context
;
; In addition, all the 'default' domains associated with a server should be
; added if incoming request filtering is desired.
; autodomain=yes
;
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no
;domain=mydomain.tld,mydomain-incoming
; Add domain and configure incoming context
; for external calls to this domain
;domain=1.2.3.4 ; Add IP address as local domain
; You can have several "domain" settings
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
; Default is yes
;autodomain=yes ; Turn this on to have Asterisk add local host
; name and local IP to domain list.
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
; non-peers, use your primary domain "identity"
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.
;------------------------------ JITTER BUFFER CONFIGURATION
--------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on
the receiving side of a
; SIP channel. Defaults to "no". An
enabled jitterbuffer will
; be used only if the sending side can
create and the receiving
; side can not accept jitter. The SIP
channel can accept jitter,
; thus a jitterbuffer on the receive SIP
side will be used only
; if it is forced and enabled.
; jbforce = no ; Forces the use of a jitterbuffer on
the receive side of a SIP
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over
which the jitterbuffer is
; resynchronized. Useful to improve the
quality of the voice, with
; big jumps in/broken timestamps,
usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on
the receiving side of a SIP
; channel. Two implementations are
currently available - "fixed"
; (with size always equals to jbmaxsize)
and "adaptive" (with
; variable size, actually the new jb of
IAX2). Defaults to fixed.
; jbtargetextra = 40 ; This option only affects the jb when
'jbimpl = adaptive' is set.
; The option represents the number of
milliseconds by which the new jitter buffer
; will pad its size. the default is 40,
so without modification, the new
; jitter buffer will set its size to the
jitter value plus 40 milliseconds.
; increasing this value may help if your
network normally has low jitter,
; but occasionally has spikes.
; jblog = no ; Enables jitterbuffer frame logging.
Defaults to "no".
;-----------------------------------------------------------------------------------
[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
; any credentials in peer/register definition if realm is matched.
;
; This way, Asterisk can authenticate for outbound calls to other
; realms. We match realm on the proxy challenge and pick an set of
; credentials from this list
; Syntax:
; auth = <user>:<secret>@<realm>
; auth = <user>#<md5secret>@<realm>
; Example:
;auth=mark:topsecret@digium.com
;
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm
;------------------------------------------------------------------------------
; DEVICE CONFIGURATION
;
; 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).
;[sip_proxy]
; 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
;context=from-fwd
;host=fwd.pulver.com
;[sip_proxy-out]
;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!
;fromdomain=provider.sip.domain
;host=box.provider.com
;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
;[provider1]
;type=peer
;host=sip.provider1.com
;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.
;
; 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
dtmfmode=rfc2833
context=from-office
type=friend
[natted-phone](!,basic-options) ; another template inheriting basic-options
nat=yes
directmedia=no
host=dynamic
[public-phone](!,basic-options) ; another template inheriting basic-options
nat=no
directmedia=yes
[my-codecs](!) ; a template for my preferred codecs
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
[ulaw-phone](!) ; and another one for ulaw-only
disallow=all
allow=ulaw
; and finally instantiate a few phones
;
; [2133](natted-phone,my-codecs)
; secret = peekaboo
; [2134](natted-phone,ulaw-phone)
; secret = not_very_secret
; [2136](public-phone,ulaw-phone)
; secret = not_very_secret_either
; ...
;
; Standard configurations not using templates look like this:
;
;[grandstream1]
;type=friend
;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=192.168.0.23 ; 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=alaw
;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
;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;type=friend
;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
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
;registertrying=yes ; Send a 100 Trying when the device registers.
;[snom]
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;secret=blah
;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=192.168.0.59 ; 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"
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;[polycom]
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;secret=blahpoly
;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
;defaultip=192.168.40.123
; Normally you do NOT need to set
this parameter
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no ; Polycom phones don't work properly
with "never"
;[pingtel]
;type=friend
;secret=blah
;host=dynamic
;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=192.168.0.60 ; IP address to use if peer has not registered
;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account
based on IP address
;permit=192.168.0.60/255.255.255.0
;permit=192.168.0.60/24 ; we can also use CIDR notation for
subnet masks
;[cisco1]
;type=friend
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to the IP address
that packet is
; received from instead of trusting
SIP headers
;host=dynamic ; This device registers with us
;directmedia=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some
devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip=192.168.0.4 ; IP address to use until registration
;defaultuser=goran ; Username to use when calling this
device before registration
; Normally you do NOT need to set
this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all
calls from or to this device
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended transfer.
;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;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.
;
[80]
type=friend
host=dynamic
language=es
context=internal
secret=80
username=80
callerid=80
dtmfmode=rfc2833
qualify=yes
;
[81]
type=friend
host=dynamic
language=es
context=internal
secret=81
username=81
callerid=81
dtmfmode=rfc2833
qualify=yes
;
Gracias!
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