Asterisk als SIP-Unteranlage am internen S=-BUS der Hicom 118
NTBA --> S0(PTMP)-HICOM118-S0(PTMP) --> Fritzcard-USB --> Asterisk --> SIP-Phone(X-Lite)
SIP-Nebenstellen im Asterisk 71,72,291,292,295,391,392,197,191
Nebenstelle in der Hicom z.B. 32,31,21,22,25...
System: Suse9.3 mit 256MB RAM auf 500MHz-System. Asterisk 1.2.6 vom März 2006.
Was bis jetzt geht: SIP-Intern nach SIP-Intern
SIP-Intern nach ISDN-Hicom118 und über Hicom ins Amt
-- Executing Dial("SIP/SIP71-6bad", "CAPI/contr1/32|30") in new stack
-- Called contr1/32
-- CAPI/ISDN1/32-39 is making progress passing it to SIP/SIP71-6bad
-- CAPI/ISDN1/32-39 is proceeding passing it to SIP/SIP71-6bad
-- CAPI/ISDN1/32-39 is ringing
-- CAPI/ISDN1/32-39 answered SIP/SIP71-6bad
-- Started music on hold, class 'default', on channel 'CAPI/ISDN1/32-39'
-- Stopped music on hold on CAPI/ISDN1/32-39
== ISDN1: CAPI Hangingup
== Spawn extension (SIP-ARN, 32, 1) exited non-zero on 'SIP/SIP71-6bad'
Was auch geht... Amt über Hicom zum Asterisk zum SIP-Client
== ISDN1: Incoming call '00362112345' -> '71'
-- Executing Dial("CAPI/ISDN1/71-38", "SIP/SIP71|30") in new stack
-- Called SIP71
-- SIP/SIP71-6a01 is ringing
> CAPI INFO 0x3490: Normal call clearing
== Spawn extension (capi-in, 71, 1) exited non-zero on 'CAPI/ISDN1/71-38'
== ISDN1: CAPI Hangingup
Was nicht geht: Ein Anruf von der Hicom über den Astrisk zum SIP-Client
== ISDN1: Incoming call '32' -> '71' ; Versuch von der Hicom Nst.32 zum Asterisk Nst.71 zu Telefonieren
== ISDN1: CAPI Hangingup
Asterisk erkennt den Ruf, macht aber nichts...
------ capi.conf ------------------------------------------------------
;
; CAPI config
;
;
; general section
[general]
nationalprefix=
internationalprefix=
rxgain=0.8
txgain=0.8
language=de ;set default language
;ulaw=yes ;set this, if you live in u-law world instead of a-law
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
msn=71
incomingmsn=* ;71,72,291,292,295,391,392,197,191
;allow incoming calls to this list of MSNs/DIDs, * = any
defaultcid=71 ;set a default caller id to that interface for dial-out,
;this caller id will be used when dial option 'd' is set.
;controller=0 ;ISDN4BSD default
;controller=7 ;ISDN4BSD USB default
controller=1 ;capi controller number to use
group=1 ;dialout group
;prefix=0 ;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
;holdtype=hold ;when Asterisk puts the call on hold, ISDN HOLD will be used. If
;set to 'local' (default value), no hold is done and Asterisk may
;play MOH.
;immediate=yes ;DID: immediate start of pbx with extension 's' if no digits were
; received on incoming call (no destination number yet)
;MSN: start pbx on CONNECT_IND and don't wait for SETUP/SENDING-COMPLETE.
; info like REDIRECTINGNUMBER may be lost, but this is necessary for
; drivers/pbx/telco which does not send SETUP or SENDING-COMPLETE.
;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)
;echotail=64 ;echo cancel tail setting
;bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;Asterisk call group
devices=2 ;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
----------------- sip.conf --------------------------------------
[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
; if asterisk was compiled with OSP support.
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=192.168.1.195 ; IP address to bind to (0.0.0.0 binds to all)
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
;domain=mydomain.tld ; Set default domain for this host
; If configured, Asterisk will only allow
; INVITE and REFER to non-local domains
; Use "sip show domains" to list local domains
;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
;allowexternalinvites=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.
;pedantic=yes ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
;tos=184 ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpiry=3600 ; Max length of incoming registration we allow
;defaultexpiry=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"
;videosupport=yes ; Turn on support for SIP video
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ;
;musicclass=default ; Sets the default music on hold class for all SIP calls
; This may also be set for individual users/peers
language=de ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we're not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
; when we're on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;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
;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 SIP 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
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
;compactheaders = yes ; send compact sip headers.
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
;notifyringing = yes ; Notify subscriptions on RINGING state
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[authentication]
;------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
;
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; permit permit
; deny deny
; secret secret
; md5secret md5secret
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
; callgroup callgroup
; pickupgroup pickupgroup
; language language
; allow allow
; disallow disallow
; insecure insecure
; trustrpid trustrpid
; progressinband progressinband
; promiscredir promiscredir
; useclientcode useclientcode
; accountcode accountcode
; setvar setvar
; callerid callerid
; amaflags amaflags
; call-limit call-limit
; restrictcid restrictcid
; subscribecontext subscribecontext
; mailbox
; username
; template
; fromdomain
; regexten
; fromuser
; host
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
; sendrpid
;#######################################################################################################################
[SIP71] ; X-Lite client 71
type=friend
username=SIP71
secret=SIP71
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="Test 71" <71>
context=SIP-ARN
[SIP72] ; X-Lite client 72
type=friend
username=SIP72
secret=SIP72
auth=md5,plaintext
nat=no ; we assume clients are behind NAT
host=dynamic ; and have dynamic IP addresses
reinvite=no ; if so, we need to make them
canreinvite=no ; always go through Asterisk
qualify=1000
dtmfmode=inband
callerid="Test 72" <72>
context=SIP-ARN ; use a context that exists ;-)
[SIP291] ; SIP-Client Sandra #291
type=friend
username=SIP291
secret=SIP291
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Sandra" <291>
context=SIP-ARN
[SIP292] ; SIP-Client Susi #292
type=friend
username=SIP292
secret=SIP292
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Susi" <292>
context=SIP-ARN
[SIP295] ; SIP-Client Sophie #295
type=friend
username=SIP295
secret=SIP295
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Sophie" <295>
context=SIP-ARN
[SIP392] ; SIP-Client Gaestezimmer #392
type=friend
username=SIP392
secret=SIP392
auth=plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Gaestezimmer" <392>
context=SIP-ARN
[SIP391] ; SIP-Client Computerecke #391
type=friend
username=SIP391
secret=SIP391
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Computerecke" <391>
context=SIP-ARN
[SIP197] ; SIP-Client Carport #197
type=friend
username=SIP197
secret=SIP197
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Carport" <197>
context=SIP-ARN
[SIP191] ; SIP-Client Stube #191
type=friend
username=SIP191
secret=SIP191
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Stube" <191>
context=SIP-ARN
-------------- extensions.conf -----------------------------------
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;
; 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
; preceeded by a one.
;
[capi-out]
[capi-in]
exten = 71,1,Dial(SIP/SIP71,30)
exten = 72,1,Dial(SIP/SIP72,30)
exten = 291,1,Dial(SIP/SIP291,30)
exten = 292,1,Dial(SIP/SIP292,30)
exten = 295,1,Dial(SIP/SIP295,30)
exten = 392,1,Dial(SIP/SIP392,30)
exten = 391,1,Dial(SIP/SIP391,30)
exten = 197,1,Dial(SIP/SIP197,30)
exten = 191,1,Dial(SIP/SIP191,30)
exten = t,1,Hangup
[local-sip]
;
;
[SIP-ARN] ; context for X-Lite clients
;
; Alle X9X-Nummern sind interne SIP-Clients
exten => _X[9]!,1,NoOp(“call for “${EXTEN})
exten => _X[9]!,2,Dial(SIP/SIP${EXTEN},60,tr)
exten => _X[9]!,3,Congestion
;
; Alles andere waehlt er ISDN-Capi auf die vorgeschaltete HICOM 118
exten =_X[0-8]!,1,Dial(CAPI/contr1/${EXTEN},30)
;
; interne SIP-Nummern
exten = 71,2,Dial(SIP/SIP71,30)
exten = 72,2,Dial(SIP/SIP72,30)
exten = 291,2,Dial(SIP/SIP291,30)
exten = 292,2,Dial(SIP/SIP292,30)
exten = 295,2,Dial(SIP/SIP295,30)
exten = 392,2,Dial(SIP/SIP392,30)
exten = 391,2,Dial(SIP/SIP391,30)
exten = 197,2,Dial(SIP/SIP197,30)
exten = 191,2,Dial(SIP/SIP191,30)
exten = t,1,Hangup
-------- dialplan -----------------------------------
*CLI> show dialplan
[ Context 'SIP-ARN' created by 'pbx_config' ]
'191' => 2. Dial(SIP/SIP191|30) [pbx_config]
'197' => 2. Dial(SIP/SIP197|30) [pbx_config]
'291' => 2. Dial(SIP/SIP291|30) [pbx_config]
'292' => 2. Dial(SIP/SIP292|30) [pbx_config]
'295' => 2. Dial(SIP/SIP295|30) [pbx_config]
'391' => 2. Dial(SIP/SIP391|30) [pbx_config]
'392' => 2. Dial(SIP/SIP392|30) [pbx_config]
'71' => 2. Dial(SIP/SIP71|30) [pbx_config]
'72' => 2. Dial(SIP/SIP72|30) [pbx_config]
't' => 1. Hangup() [pbx_config]
'_X[0-8]!' => 1. Dial(CAPI/contr1/${EXTEN}|30) [pbx_config]
'_X[9]!' => 1. NoOp(call for ${EXTEN}) [pbx_config]
2. Dial(SIP/SIP${EXTEN}|60|tr) [pbx_config]
3. Congestion() [pbx_config]
[ Context 'local-sip' created by 'pbx_config' ]
[ Context 'capi-in' created by 'pbx_config' ]
'191' => 1. Dial(SIP/SIP191|30) [pbx_config]
'197' => 1. Dial(SIP/SIP197|30) [pbx_config]
'291' => 1. Dial(SIP/SIP291|30) [pbx_config]
'292' => 1. Dial(SIP/SIP292|30) [pbx_config]
'295' => 1. Dial(SIP/SIP295|30) [pbx_config]
'391' => 1. Dial(SIP/SIP391|30) [pbx_config]
'392' => 1. Dial(SIP/SIP392|30) [pbx_config]
'71' => 1. Dial(SIP/SIP71|30) [pbx_config]
'72' => 1. Dial(SIP/SIP72|30) [pbx_config]
't' => 1. Hangup() [pbx_config]
[ Context 'capi-out' created by 'pbx_config' ]
[ Context 'parkedcalls' created by 'res_features' ]
'700' => 1. Park() [res_features]
-= 23 extensions (25 priorities) in 5 contexts. =-
*CLI>
NTBA --> S0(PTMP)-HICOM118-S0(PTMP) --> Fritzcard-USB --> Asterisk --> SIP-Phone(X-Lite)
SIP-Nebenstellen im Asterisk 71,72,291,292,295,391,392,197,191
Nebenstelle in der Hicom z.B. 32,31,21,22,25...
System: Suse9.3 mit 256MB RAM auf 500MHz-System. Asterisk 1.2.6 vom März 2006.
Was bis jetzt geht: SIP-Intern nach SIP-Intern
SIP-Intern nach ISDN-Hicom118 und über Hicom ins Amt
-- Executing Dial("SIP/SIP71-6bad", "CAPI/contr1/32|30") in new stack
-- Called contr1/32
-- CAPI/ISDN1/32-39 is making progress passing it to SIP/SIP71-6bad
-- CAPI/ISDN1/32-39 is proceeding passing it to SIP/SIP71-6bad
-- CAPI/ISDN1/32-39 is ringing
-- CAPI/ISDN1/32-39 answered SIP/SIP71-6bad
-- Started music on hold, class 'default', on channel 'CAPI/ISDN1/32-39'
-- Stopped music on hold on CAPI/ISDN1/32-39
== ISDN1: CAPI Hangingup
== Spawn extension (SIP-ARN, 32, 1) exited non-zero on 'SIP/SIP71-6bad'
Was auch geht... Amt über Hicom zum Asterisk zum SIP-Client
== ISDN1: Incoming call '00362112345' -> '71'
-- Executing Dial("CAPI/ISDN1/71-38", "SIP/SIP71|30") in new stack
-- Called SIP71
-- SIP/SIP71-6a01 is ringing
> CAPI INFO 0x3490: Normal call clearing
== Spawn extension (capi-in, 71, 1) exited non-zero on 'CAPI/ISDN1/71-38'
== ISDN1: CAPI Hangingup
Was nicht geht: Ein Anruf von der Hicom über den Astrisk zum SIP-Client
== ISDN1: Incoming call '32' -> '71' ; Versuch von der Hicom Nst.32 zum Asterisk Nst.71 zu Telefonieren
== ISDN1: CAPI Hangingup
Asterisk erkennt den Ruf, macht aber nichts...
------ capi.conf ------------------------------------------------------
;
; CAPI config
;
;
; general section
[general]
nationalprefix=
internationalprefix=
rxgain=0.8
txgain=0.8
language=de ;set default language
;ulaw=yes ;set this, if you live in u-law world instead of a-law
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
msn=71
incomingmsn=* ;71,72,291,292,295,391,392,197,191
;allow incoming calls to this list of MSNs/DIDs, * = any
defaultcid=71 ;set a default caller id to that interface for dial-out,
;this caller id will be used when dial option 'd' is set.
;controller=0 ;ISDN4BSD default
;controller=7 ;ISDN4BSD USB default
controller=1 ;capi controller number to use
group=1 ;dialout group
;prefix=0 ;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
;holdtype=hold ;when Asterisk puts the call on hold, ISDN HOLD will be used. If
;set to 'local' (default value), no hold is done and Asterisk may
;play MOH.
;immediate=yes ;DID: immediate start of pbx with extension 's' if no digits were
; received on incoming call (no destination number yet)
;MSN: start pbx on CONNECT_IND and don't wait for SETUP/SENDING-COMPLETE.
; info like REDIRECTINGNUMBER may be lost, but this is necessary for
; drivers/pbx/telco which does not send SETUP or SENDING-COMPLETE.
;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)
;echotail=64 ;echo cancel tail setting
;bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;Asterisk call group
devices=2 ;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
----------------- sip.conf --------------------------------------
[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
; if asterisk was compiled with OSP support.
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=192.168.1.195 ; IP address to bind to (0.0.0.0 binds to all)
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
;domain=mydomain.tld ; Set default domain for this host
; If configured, Asterisk will only allow
; INVITE and REFER to non-local domains
; Use "sip show domains" to list local domains
;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
;allowexternalinvites=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.
;pedantic=yes ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
;tos=184 ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpiry=3600 ; Max length of incoming registration we allow
;defaultexpiry=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to "asterisk"
;videosupport=yes ; Turn on support for SIP video
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ;
;musicclass=default ; Sets the default music on hold class for all SIP calls
; This may also be set for individual users/peers
language=de ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we're not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
; when we're on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;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
;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 SIP 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
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
;compactheaders = yes ; send compact sip headers.
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
;notifyringing = yes ; Notify subscriptions on RINGING state
disallow=all
allow=ulaw
allow=alaw
allow=gsm
[authentication]
;------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
;
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; permit permit
; deny deny
; secret secret
; md5secret md5secret
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
; callgroup callgroup
; pickupgroup pickupgroup
; language language
; allow allow
; disallow disallow
; insecure insecure
; trustrpid trustrpid
; progressinband progressinband
; promiscredir promiscredir
; useclientcode useclientcode
; accountcode accountcode
; setvar setvar
; callerid callerid
; amaflags amaflags
; call-limit call-limit
; restrictcid restrictcid
; subscribecontext subscribecontext
; mailbox
; username
; template
; fromdomain
; regexten
; fromuser
; host
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
; sendrpid
;#######################################################################################################################
[SIP71] ; X-Lite client 71
type=friend
username=SIP71
secret=SIP71
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="Test 71" <71>
context=SIP-ARN
[SIP72] ; X-Lite client 72
type=friend
username=SIP72
secret=SIP72
auth=md5,plaintext
nat=no ; we assume clients are behind NAT
host=dynamic ; and have dynamic IP addresses
reinvite=no ; if so, we need to make them
canreinvite=no ; always go through Asterisk
qualify=1000
dtmfmode=inband
callerid="Test 72" <72>
context=SIP-ARN ; use a context that exists ;-)
[SIP291] ; SIP-Client Sandra #291
type=friend
username=SIP291
secret=SIP291
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Sandra" <291>
context=SIP-ARN
[SIP292] ; SIP-Client Susi #292
type=friend
username=SIP292
secret=SIP292
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Susi" <292>
context=SIP-ARN
[SIP295] ; SIP-Client Sophie #295
type=friend
username=SIP295
secret=SIP295
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Sophie" <295>
context=SIP-ARN
[SIP392] ; SIP-Client Gaestezimmer #392
type=friend
username=SIP392
secret=SIP392
auth=plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Gaestezimmer" <392>
context=SIP-ARN
[SIP391] ; SIP-Client Computerecke #391
type=friend
username=SIP391
secret=SIP391
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Computerecke" <391>
context=SIP-ARN
[SIP197] ; SIP-Client Carport #197
type=friend
username=SIP197
secret=SIP197
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Carport" <197>
context=SIP-ARN
[SIP191] ; SIP-Client Stube #191
type=friend
username=SIP191
secret=SIP191
auth=md5,plaintext
nat=no
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Stube" <191>
context=SIP-ARN
-------------- extensions.conf -----------------------------------
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;
; 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
; preceeded by a one.
;
[capi-out]
[capi-in]
exten = 71,1,Dial(SIP/SIP71,30)
exten = 72,1,Dial(SIP/SIP72,30)
exten = 291,1,Dial(SIP/SIP291,30)
exten = 292,1,Dial(SIP/SIP292,30)
exten = 295,1,Dial(SIP/SIP295,30)
exten = 392,1,Dial(SIP/SIP392,30)
exten = 391,1,Dial(SIP/SIP391,30)
exten = 197,1,Dial(SIP/SIP197,30)
exten = 191,1,Dial(SIP/SIP191,30)
exten = t,1,Hangup
[local-sip]
;
;
[SIP-ARN] ; context for X-Lite clients
;
; Alle X9X-Nummern sind interne SIP-Clients
exten => _X[9]!,1,NoOp(“call for “${EXTEN})
exten => _X[9]!,2,Dial(SIP/SIP${EXTEN},60,tr)
exten => _X[9]!,3,Congestion
;
; Alles andere waehlt er ISDN-Capi auf die vorgeschaltete HICOM 118
exten =_X[0-8]!,1,Dial(CAPI/contr1/${EXTEN},30)
;
; interne SIP-Nummern
exten = 71,2,Dial(SIP/SIP71,30)
exten = 72,2,Dial(SIP/SIP72,30)
exten = 291,2,Dial(SIP/SIP291,30)
exten = 292,2,Dial(SIP/SIP292,30)
exten = 295,2,Dial(SIP/SIP295,30)
exten = 392,2,Dial(SIP/SIP392,30)
exten = 391,2,Dial(SIP/SIP391,30)
exten = 197,2,Dial(SIP/SIP197,30)
exten = 191,2,Dial(SIP/SIP191,30)
exten = t,1,Hangup
-------- dialplan -----------------------------------
*CLI> show dialplan
[ Context 'SIP-ARN' created by 'pbx_config' ]
'191' => 2. Dial(SIP/SIP191|30) [pbx_config]
'197' => 2. Dial(SIP/SIP197|30) [pbx_config]
'291' => 2. Dial(SIP/SIP291|30) [pbx_config]
'292' => 2. Dial(SIP/SIP292|30) [pbx_config]
'295' => 2. Dial(SIP/SIP295|30) [pbx_config]
'391' => 2. Dial(SIP/SIP391|30) [pbx_config]
'392' => 2. Dial(SIP/SIP392|30) [pbx_config]
'71' => 2. Dial(SIP/SIP71|30) [pbx_config]
'72' => 2. Dial(SIP/SIP72|30) [pbx_config]
't' => 1. Hangup() [pbx_config]
'_X[0-8]!' => 1. Dial(CAPI/contr1/${EXTEN}|30) [pbx_config]
'_X[9]!' => 1. NoOp(call for ${EXTEN}) [pbx_config]
2. Dial(SIP/SIP${EXTEN}|60|tr) [pbx_config]
3. Congestion() [pbx_config]
[ Context 'local-sip' created by 'pbx_config' ]
[ Context 'capi-in' created by 'pbx_config' ]
'191' => 1. Dial(SIP/SIP191|30) [pbx_config]
'197' => 1. Dial(SIP/SIP197|30) [pbx_config]
'291' => 1. Dial(SIP/SIP291|30) [pbx_config]
'292' => 1. Dial(SIP/SIP292|30) [pbx_config]
'295' => 1. Dial(SIP/SIP295|30) [pbx_config]
'391' => 1. Dial(SIP/SIP391|30) [pbx_config]
'392' => 1. Dial(SIP/SIP392|30) [pbx_config]
'71' => 1. Dial(SIP/SIP71|30) [pbx_config]
'72' => 1. Dial(SIP/SIP72|30) [pbx_config]
't' => 1. Hangup() [pbx_config]
[ Context 'capi-out' created by 'pbx_config' ]
[ Context 'parkedcalls' created by 'res_features' ]
'700' => 1. Park() [res_features]
-= 23 extensions (25 priorities) in 5 contexts. =-
*CLI>
