Config of SIP TRUNK
Posted: Mon Mar 02, 2015 3:00 pm
Hi,
im new with asterisk and i try to setup connection to SIP provider, i got technical information prom provider:
For SIP: IP XX.XXX.XXX.XXX Port: 5060
Data for medium/RTP: IP: XX.XXX.XXX.XXX Port: 25000-30000
Codecs: G.723.1, G.729, G.711
DTMF: INBAND,RFC2833
CLIR: Sip privacy RFC3325
Phone format: Phone. URI
And i try to setup only outbound calls:
Registration String: I leave empty becouse there is no username and password for registration
[p2sip]
disallow=all
allow=g729
allow=gsm
allow=ulaw
type=friend
port=5060
host=XX.XXX.XXX.XXX
dtmfmode=rfc2833
qualify=yes
insecure=invite
canreinvite=yes
nat=auto
ignoresdpversion=yes
context=from-trunk
Globals string: P2SIP = SIP/p2sip
Dialplan:
exten => _1348XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1348XXXXXXXXX,n,Dial(${P2SIP}/${EXTEN:4},,to)
exten => _1348XXXXXXXXX,n,Hangup
ERROR:
Call Rejected: CHANUNAVAIL
Cause: 20 - Subscriber absent.
My Asterisk:
VERSION: 2.6-374a
BUILD: 120820-1104
im new with asterisk and i try to setup connection to SIP provider, i got technical information prom provider:
For SIP: IP XX.XXX.XXX.XXX Port: 5060
Data for medium/RTP: IP: XX.XXX.XXX.XXX Port: 25000-30000
Codecs: G.723.1, G.729, G.711
DTMF: INBAND,RFC2833
CLIR: Sip privacy RFC3325
Phone format: Phone. URI
And i try to setup only outbound calls:
Registration String: I leave empty becouse there is no username and password for registration
[p2sip]
disallow=all
allow=g729
allow=gsm
allow=ulaw
type=friend
port=5060
host=XX.XXX.XXX.XXX
dtmfmode=rfc2833
qualify=yes
insecure=invite
canreinvite=yes
nat=auto
ignoresdpversion=yes
context=from-trunk
Globals string: P2SIP = SIP/p2sip
Dialplan:
exten => _1348XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1348XXXXXXXXX,n,Dial(${P2SIP}/${EXTEN:4},,to)
exten => _1348XXXXXXXXX,n,Hangup
ERROR:
Call Rejected: CHANUNAVAIL
Cause: 20 - Subscriber absent.
My Asterisk:
VERSION: 2.6-374a
BUILD: 120820-1104