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!