Razvan Cosma
2005-Apr-04 02:22 UTC
[Asterisk-Users] Asterisk+Sipgate - just one step away..
Hello all,
I have a working Asterisk setup, also a working sipgate.co.uk account
(tested with a GrandStream ATA 486), but got stuck in forwarding calls
from local users to sipgate. Very frustrating, since I feel there's just
one silly error somewhere.. story follows:
REGISTER both of the local user to * and of the * to sipgate.co.uk is
successful
but when dialing some random phone number in Linphone in the form
sip:111111111@1.2.3.4 (1.2.3.4 is the * box) I get
-- Executing SetCallerID("SIP/user-733d",
"xxxxxxxx@sipgate.co.uk")
in new stack
-- Executing Dial("SIP/user-733d",
"SIP/1111111111@sipgate.co.uk|30|tr") in new stack
Outgoing Call for 1111111111111111
111111111111 is not a local user
-- Called 11111111111@sipgate.co.uk
Failed to authenticate on INVITE to ''xxxxxxx@sipgate.co.uk"
<sip:yyyyyyyy@1.2.3.4>;tag=as319c47a2'
^^^^ this I think is the problem - while the call is redirected, the
correct number is dialed, Asterisk says it changed the callerid, but
"yyyyyy" is the local username and "1.2.3.4" is the *
address, shouldnt'
it be ''xxxxxxx@sipgate.co.uk" ?
sip.conf:
[general]
register => xxxxxxxx:ppppppp@sipgate.co.uk/xxxxxx
[sipgate]
type=peer
username=xxxxxxxx
secret=pppppppppp
host=sipgate.co.uk
fromuser=xxxxxxxx
fromdomain=sipgate.co.uk
nat=no
authuser=xxxxxxxx
dtmfmode=info
context=incomingsipgate
context=default
insecure=very
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
extensions.conf:
[general]
static=yes
writeprotect=yes
[incomingsipgate]
exten => h,1,Hangup
exten => xxxxxxx,1,Dial(SIP/102,20,tr)
[sipgate]
exten => _9.,1,SetCallerID(xxxxxxx@sipgate.co.uk)
exten => _9.,2,Dial(SIP/${EXTEN:1}@sipgate.co.uk,30,tr)
exten => _9.,3,Playback(invalid)
exten => _9.,4,Hangup
Any hints please?
Thank you very much
Razvan
administrator tootai
2005-Apr-04 02:46 UTC
[Asterisk-Users] Asterisk+Sipgate - just one step away..
Razvan Cosma a ?crit :> Hello all, > I have a working Asterisk setup, also a working sipgate.co.uk account > (tested with a GrandStream ATA 486), but got stuck in forwarding calls > from local users to sipgate. Very frustrating, since I feel there's > just one silly error somewhere.. story follows: > REGISTER both of the local user to * and of the * to sipgate.co.uk is > successful > but when dialing some random phone number in Linphone in the form > sip:111111111@1.2.3.4 (1.2.3.4 is the * box) I get > > -- Executing SetCallerID("SIP/user-733d", "xxxxxxxx@sipgate.co.uk") > in new stack > -- Executing Dial("SIP/user-733d", > "SIP/1111111111@sipgate.co.uk|30|tr") in new stack > Outgoing Call for 1111111111111111 > 111111111111 is not a local user > -- Called 11111111111@sipgate.co.uk > Failed to authenticate on INVITE to ''xxxxxxx@sipgate.co.uk" > <sip:yyyyyyyy@1.2.3.4>;tag=as319c47a2' > ^^^^ this I think is the problem - while the call is redirected, the > correct number is dialed, Asterisk says it changed the callerid, but > "yyyyyy" is the local username and "1.2.3.4" is the * address, > shouldnt' it be ''xxxxxxx@sipgate.co.uk" ? > > sip.conf: > [general] > register => xxxxxxxx:ppppppp@sipgate.co.uk/xxxxxx > [sipgate] > type=peer > username=xxxxxxxx > secret=pppppppppp > host=sipgate.co.uk > fromuser=xxxxxxxx > fromdomain=sipgate.co.uk > nat=no > authuser=xxxxxxxx > dtmfmode=info > context=incomingsipgate > context=default > insecure=very > canreinvite=yes > disallow=all > allow=ulaw > allow=alaw > > extensions.conf: > [general] > static=yes > writeprotect=yes > [incomingsipgate] > exten => h,1,Hangup > exten => xxxxxxx,1,Dial(SIP/102,20,tr) > [sipgate] > exten => _9.,1,SetCallerID(xxxxxxx@sipgate.co.uk) > exten => _9.,2,Dial(SIP/${EXTEN:1}@sipgate.co.uk,30,tr) > exten => _9.,3,Playback(invalid) > exten => _9.,4,Hangup > > > Any hints please?according to your sip.conf, should be [...] exten => _9.,2,Dial(SIP/${EXTEN:1}@sipgate,30,tr) in extensions.conf
Razvan Cosma
2005-Apr-04 03:08 UTC
[Asterisk-Users] Asterisk+Sipgate - just one step away..
On 04/04/2005 12:46 PM, administrator tootai wrote:> > according to your sip.conf, should be > [...] > exten => _9.,2,Dial(SIP/${EXTEN:1}@sipgate,30,tr) > > in extensions.conf >Yessss :) Thank you very much!