On 11/28/06, Bogdan-Andrei Iancu <bogdan@voice-system.ro> wrote:

Hi Sam,

According to SIP RFC the TO header is not used at all for routing - most
probably you have an old gateway which is not SIP compliant anymore.
There is no mechanism in openser to change the TO header. The strip()
function affects only the RURI.

the authentication name *must* not be changed as the auth will failed -
the auth response is computed based on the auth name known by the UAC.

regards,
bogdan

Sam Hsu wrote:

> i use the strip() function to strip the prefix when call out
> for example, prefix 0 to call out
> there are my sip invite message
> ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
>
> INVITE sip:0928117466@202.111.222.76:5060 SIP/2.0
> Record-Route: <sip:211.111.222.102
> <http://211.111.222.102 >;r2=on;lr=on;ftag=694064c4>
> Record-Route: <sip:211.111.222.102
> <http://211.111.222.102>;transport=tcp;r2=on;lr=on;ftag=694064c4>
> Content-Length: 324
> Content-Type: application/sdp
> Via: SIP/2.0/UDP 211.23.176.102
> <http://211.23.176.102>;branch= z9hG4bK4eec.ecf680e4.0;i=1
> Via: SIP/2.0/TCP 192.168.123.5:5060
> <http://192.168.123.5:5060>;received= 220.132.138.7
> <http://220.132.138.7>;branch=z9hG4bK69486617
> To: <sip:00928117466@211.111.222.102
> <mailto: sip:00928117466@211.111.222.102>>
> From: "joepass" <sip:joepass@211.111.222.102
> <mailto:sip:joepass@211.111.222.102>>;tag=694064c4
> Supported: timer
> Call-ID: 95746504-39004973-1533c430-d8319d29@192.168.123.5
> <mailto:95746504-39004973-1533c430-d8319d29@192.168.123.5>
> CSeq: 26589 INVITE
> User-Agent: IP SIP Phone/2.1.3
> Max-Forwards: 69
> Session-Expires: 1800
> Allow: UPDATE,INFO,MESSAGE,REFER,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL
> Authorization: Digest
> nonce="456ba868fbd62c72ca16fcdd04678168a8fa0683", username="joepass",
> realm="votel-tech.com <http://votel-tech.com>", uri="
> sip:00928117466@211.111.222.102
> <mailto:sip:00928117466@211.111.222.102>",
> response="73a8c869c2a42a12f0d920c2a7d6f068"
> P-IPRAuth: votel-tech.com <http://votel-tech.com>
> Contact: <sip:joepass@220.111.222.7:1070>
> ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
>
> in the example, the real number is 0928117466
> add prefix 0, so the final number is 00928117466
> I see the INVITE part the id has strip the prefix 0
> but
> To: < sip:00928117466@211.111.222.102
> <mailto:sip:00928117466@211.111.222.102>>
> and the Authorization part(uri) still keep on 00928117466
>
> My gateway seems to use this information to call out.
> So it cause some error respond.
> If i call to the gateway directly(not through openser with number
> 0928117466), it works.
> How can i strip the "To" and "Authorization" part uri.
> thanks...
>
> Sam
>
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