Hello there
I am new to Asterisk, so bear with me, i think i understand the flow, but i am not completely sure, i think i am on the right track.
devices
1 Grandstream SIP/100
2 Zoiper softphone SIP/101
Ok, we are using CentOs 5.4 freshly installed, and Asterisk installed with yum repos.
We are using a SIP trunk. To make incoming and external calls.
I’ve got 2 problems.
1, is that our main number doesn’t show up on my mobile phone when i make an external call from my Grandstream SIP phone.
which is registred on my asterisk, i get a hidden number showing up on my mobile phone.
With Zoiper softphone i get a number which don’t belong to us, and when i dial that number back again, i get an answering machine who says that this number is not in use, and it is used for “commercial” use when numbers is not registred, this is ok.
2, when i try to call our main number, we get busy sound. When i say this, i mean external calls in to our Asterisk.
[Apr 30 11:28:37] NOTICE[7360]: chan_sip.c:20059 handle_request_invite: Call from ‘’ to extension ‘4767209610’ rejected because extension not found.
the number 4767209610 is our main number, our public number, this is btw in Norway +47 is our country code.
So the external call is handled by the Asterisk, but it cant put it trough to SIP/100, my grandstream.
SIP.conf:
[code][general]
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
;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)
useragent=Phonect_Asterisk ; Fra Phonect Manual
Register => 47xxxxxxxx:mypasswd@odin.service.ipallover.net
[phonect_01]
type=peer
context=default
username=47xxxxxxxx
secret=mypasswd
host=domain.name.from.SIP.provider
useragent=Phonect_Asterisk
canrenvite=no
disallow=all
allow=alaw
allow=ulaw
; Abacus test entries
[100]
type=friend
username=100
secret=1234
host=dynamic
context=abacus_test
disallow=all
allow=ulaw
allow=alaw
[roald]
type=friend
username=roald
secret=1234
host=dynamic
context=abacus_test
[/code]
Abacus_test is what i want to test.
in sip.conf
srvlookup=yes is further down the file.
extensions.conf
[code][globals]
SIPTRUNK1=SIP/phonect_01 ; SIP trunk fra Phonect
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
;Abacus test extension
[abacus_test]
exten => s,1,Noop(Extract DID from SIP TO header )
exten => s,2,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2):2},1)
; Direkte internnummer
exten => 100,1,Dial(SIP/100)
exten => 101,1,Dial(SIP/roald)
; Tast 0+nummer for å ringe eksternt
exten => _0X.,1,NoOp(Ringer til nr ${EXTEN:1})
exten => _0X.,n,Dial(${SIPTRUNK1}/${EXTEN:1})
exten => _0X.,n,Goto(${DIALSTATUS})
exten => _0X.,n(BUSY),Busy
exten => _0X.,n(NOANSWER),Hangup
[from-trunk]
; Eksterne, inngående samtaler
exten => 67209610,1,Dial(SIP/100)
exten => 67209611,1,Dial(SIP/101)
[/code]
there is some norwegian language and comments in the .conf file, but i think u understand.
So, i hope anyone out there could help me understand why i can’t make external callsin to asterisk and my sip/100 grandstream, and why our public phone number 67209610 won’t show when i’m making external calls from my grandstream.
Thanks in advance!
Aksel