Hallo, vielen Dank erstmal für das informative Forum hier, hab schon so einiges interessantes gelesen.
Ich habe mit meinem Asterisk so ein kleines Problem, ich habe mir im Internet (was nicht gerade von vorteil ist) ein kleine
config zusammengebastelt, leider funktioniert das mit dem Raustelefonieren nicht.
Ich muss sagen ich bin eine totaler noob was Asterisk angeht.
Ich möchte gerne mit dem Telefon 5010 über Sipgate raustelefonieren doch leider erhalte ich nachstehende Fehlermeldungen.
Asterisk 1.2.13
gw*CLI> sip show registry
Host Username Refresh State
sipgate.de:5060
Auszug asterisk -vvvr
sip.conf
extensions.conf
Wäre dankbar wenn mir jemand helfen könnte
Ich habe mit meinem Asterisk so ein kleines Problem, ich habe mir im Internet (was nicht gerade von vorteil ist) ein kleine
config zusammengebastelt, leider funktioniert das mit dem Raustelefonieren nicht.
Ich muss sagen ich bin eine totaler noob was Asterisk angeht.
Ich möchte gerne mit dem Telefon 5010 über Sipgate raustelefonieren doch leider erhalte ich nachstehende Fehlermeldungen.
Asterisk 1.2.13
gw*CLI> sip show registry
Host Username Refresh State
sipgate.de:5060
Auszug asterisk -vvvr
Code:
Verbosity is at least 5
-- Executing Dial("SIP/5010-08192438", "SIP/6658@8776847e2") in new stack
Apr 2 10:18:48 WARNING[11786]: chan_sip.c:2017 create_addr: No such host: 8776847e2
Apr 2 10:18:48 NOTICE[11786]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Playback("SIP/5010-08192438", "invalid") in new stack
-- Playing 'invalid' (language 'de')
-- Executing Hangup("SIP/5010-08192438", "") in new stack
== Spawn extension (outgoing_sipgate, 740305, 3) exited non-zero on 'SIP/5010-08192438'
-- Executing Dial("SIP/5010-08192438", "SIP/072566658@8776847e2") in new stack
Apr 2 10:19:04 WARNING[11790]: chan_sip.c:2017 create_addr: No such host: 8776847e2
Apr 2 10:19:04 NOTICE[11790]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Playback("SIP/5010-08192438", "invalid") in new stack
-- Playing 'invalid' (language 'de')
-- parse_srv: SRV mapped to host sipgate.de, port 5060sip.conf
Code:
[general]
; all comments refer to lines above them.
context = default
port = 5060
bindaddr = 192.168.1.254
srvlookup=yes
language=de
; bind to all addresses - allows registration to Sipgate via
; internet-facing interface, and phones to register on
; LAN-facing interface. YMMV, depending on network topology
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g729
allow=gsm
allow=all
; allowed codecs
localnet=192.168.1.0/24
; Just yer standard LAN interface
register => 8776....:[email protected]/8776....
; replace 1234567 with your sipgate ID
[sipgate]
type=peer
context=incoming_sipgate
fromuser=8776....
username=8776....
;authuser=1234567
; not sure if its really necessary 3 times?
secret=PASSW
host=sipgate.de
fromdomain=sipgate.de
insecure=very
qualify=yes
canreinvite=no
nat=no
;net=yes
;disallow=all
;allow=ulaw
;allow=alaw
[5010]
type=friend
context=outgoing_sipgate
; this extension can make calls via outgoing_sipgate in extensions.conf.
; Just comment it out to disable outgoing calls from this extension.
username=5010
secret=9876
dtmfmode=rfc2833
callerid=Loft <5010>
mailbox=5010
disallow=all
allow=ulaw
allow=alaw
qualify=yes
host=dynamic
[5012]
type=friend
context=outgoing_sipgate
; can place calls via outgoing_sipgate
username=5012
secret=9876
dtmfmode=rfc2833
callerid=Cordless <5012>
mailbox=5012
; note they're all sharing one mailbox
disallow=all
allow=ulaw
allow=alaw
qualify=yes
host=dynamicextensions.conf
Code:
[general]
static=yes
writeprotect=no
[default]
; the reason i break these out in to separate [sections]
; is so that they can be excluded/included by commenting out
; one line. Makes debugging a lot easier.
include => util_numbers
include => internal_phones
include => voicemailboxes
include => outgoing_sipgate
[internal_phones]
;My internal phones
exten => 5010,1,Dial(SIP/5010,530)
exten => 5010,2,Voicemail(u5010)
; SIP Hardphone 5010: rings for 50 seconds, then
; goes to voicemailbox 5010
;exten => 5012,1,Dial(SIP/5012,30)
;exten => 5012,2,Voicemail(u5012)
; SIP ATA 5012: rings for 30 seconds, then
; goes to voicemailbox 5012
[voicemailboxes]
;main mailbox menu
exten => 5000,1,VoicemailMain
exten => 5000,2,Hangup
; dial 5000 for voicemail
[outgoing_sipgate]
;include => default
; dial 9 for an outside line (via Sipgate)
; needs to be in its own context for security porpoises
exten => _X.,1,Dial(SIP/${EXTEN}@8776....)
exten => _X.,2,Playback(invalid)
exten => _X.,3,Hangup
;sipgate stuff...
[daytime_sipgate]
; replace 1234567 with your sipgate ID
exten => 8776....,1,NoOp(--- ${CALLERID} calling on sipgate (${EXTEN}) ---)
exten => 8776....,2,Dial(SIP/5010&SIP/5012,30)
; extensions 5010 and 5012 ring on incoming calls from sipgate.
; Goes to voicemail after 30 seconds
exten => 8776....,3,Answer
exten => 8776....,4,Wait,1
exten => 8776....,5,Voicemail(u5010)
; voicemail goes to 5010's mailbox
exten => 8776....,6,hangup
[night_sipgate]
; diverts all calls to voicemail
exten => 8776....,1,NoOp(--- ${CALLERID} calling on sipgate (${EXTEN}) ---)
exten => 8776....,2,Answer
exten => 8776....,3,Wait,1
; need to record night message and play here
exten => 8776....,4,Voicemail(u5010)
exten => 8776....,5,hangup
[incoming_sipgate]
include => daytime_sipgate|8:00-20:00
; between 0700 and 2300, uses daytime_sipgate
include => night_sipgate|20:01-7:59
; between 2301 and 0659, use night_sipgate.
; Can specify days/weeks/months also.
; utility numbers
; various utilities, as a demo of asterisk's capabilities.
; Probably irrelevant in this digital telephony age.
[util_numbers]
include => onesevenohsevenoh
include => onetwothree
; this bit is here so that one only has to include => one thing
; up in [default] ^
[onesevenohsevenoh]
; 17070 is bt's line test facility, reads out your number
; we don't need line test, but this is here to say what extension is calling
; pointless but nice
exten => 17070,1,Answer
exten => 17070,2,Wait(1)
exten => 17070,3,SayDigits(${CALLERIDNUM})
exten => 17070,4,Hangup
[onetwothree]
; 123 is the uk speaking clock
exten => 123,1,Answer
exten => 123,2,Wait(1)
exten => 123,3,SayUnixTime( | | k)
exten => 123,4,SayUnixTime( | | M)
exten => 123,5,Playback(vm-and)
exten => 123,6,SayUnixTime( | | S)
exten => 123,7,Playback(seconds)
; downloaded seconds.gsm [amongst others] from
; http://www.loligo.com/asterisk/sounds/
; tip: search google with filetype:gsm to find .gsm files
exten => 123,8,Hangup
