Re: [Linphone-users] Linphone and asterix

[Stephen L Johnson]

On Wed, 2004-03-17 at 09:19, Simon MORLAT wrote:
> Hello,
> 
> The problem is that you did not supply a password to linphone, so it can 
> build the authentified request.
> 
> Simon

Would this password be the same password used in for SIP Registrar
Server? I do have a password entered for the SIP Registrar.



> 
> Stephen L Johnson wrote:
> 
> >I'm trying to connect with an asterisk server,  I can get Linphone to
> >REGISTER with successfully, but I can't get a call to connect. The
> >response code I get is "User needs to authenticate to INVITE." 
> >
> >>From the mail archives I see a similar issue, but the problem was due to
> >the Contact: field not being filled by asterisk. As you can see from my
> >call log before, that is isn't the problem. The same error persists no
> >matter how the call is placed, direct or SIP proxy
> >
> >
> >Linphone -> asterisk call log 
> >----------------------
> > INFO1 | <udp.c: 295> Sending message:
> >INVITE sip:address@hidden SIP/2.0
> >Via: SIP/2.0/UDP 208.191.248.4:5060;branch=z9hG4bK1220382814
> >From: <sip:address@hidden>;tag=304790870;tag=1386613372
> >To: <sip:address@hidden>
> >Call-ID: address@hidden
> >CSeq: 20 INVITE
> >Contact: <sip:address@hidden>
> >max-forwards: 10
> >user-agent: oSIP/Linphone-0.12.1
> >Content-Type: application/sdp
> >Content-Length:   375
> > 
> >v=0
> >o=sjohnson 123456 654321 IN IP4 208.191.248.4
> >s=A conversation
> >c=IN IP4 208.191.248.4
> >t=0 0
> >m=audio 7078 RTP/AVP 0 3 8 110 111 115 101
> >b=AS:110 20
> >b=AS:111 28
> >a=rtpmap:0 PCMU/8000/1
> >a=rtpmap:3 GSM/8000/1
> >a=rtpmap:8 PCMA/8000/1
> >a=rtpmap:110 speex/8000/1
> >a=rtpmap:111 speex/16000/1
> >a=rtpmap:115 1015/8000/1
> >a=rtpmap:101 telephone-event/8000
> >a=fmtp:101 0-11
> > 
> >| INFO1 | <udp.c: 193> info: Message from [208.191.248.3]:5060
> > 
> >| INFO1 | <udp.c: 206> info: RECEIVING UDP MESSAGE:
> >SIP/2.0 407 Proxy Authentication Required
> >Via: SIP/2.0/UDP 208.191.248.4:5060;branch=z9hG4bK1220382814
> >From: <sip:address@hidden>;tag=304790870;tag=1386613372
> >To: <sip:address@hidden>;tag=as53a16746
> >Call-ID: address@hidden
> >CSeq: 20 INVITE
> >User-Agent: Asterisk PBX
> >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> >Contact: <sip:address@hidden>
> >Proxy-Authenticate: Digest realm="asterisk", nonce="66d02411"
> >Content-Length: 0
> > 
> > 
> > 
> >| INFO1 | <udp.c: 295> Sending message:
> >ACK sip:address@hidden SIP/2.0
> >Via: SIP/2.0/UDP 208.191.248.4:5060;branch=z9hG4bK1220382814
> >From: <sip:address@hidden>;tag=304790870;tag=1386613372
> >To: <sip:address@hidden>;tag=as53a16746
> >Call-ID: address@hidden
> >CSeq: 20 ACK
> >Content-Length: 0
> > 
> > 
> >| INFO1 | <ict_callbacks.c: 233> OnEvent_New_Incoming4xxResponse!
> > 
> >| INFO1 | <ict_callbacks.c: 261> User need to authenticate to INVITE!
> > 
> >| INFO1 | <osipdialog.c: 754> password = (null)
> > 
> >| INFO1 | <authentication.c: 351> Unable to get a password: no
> >registration context.
> >| INFO1 | <osipdialog.c: 832> error: could not build the authorization
> >header.
> > 
> >| INFO1 | <ict_callbacks.c: 30> Transaction 2 killed.
> > 
> >| INFO1 | <ict_callbacks.c: 30> Transaction 3 killed.
> >  
> >