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;r2=on;lr=on;ftag=694064c4 Record-Route: sip: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;branch=z9hG4bK4eec.ecf680e4.0;i=1 Via: SIP/2.0/TCP 192.168.123.5:5060;received=220.132.138.7 ;branch=z9hG4bK69486617 To: sip:00928117466@211.111.222.102 From: "joepass" sip:joepass@211.111.222.102;tag=694064c4 Supported: timer Call-ID: 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", uri=" sip:00928117466@211.111.222.102", response="73a8c869c2a42a12f0d920c2a7d6f068" P-IPRAuth: 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 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
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
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Thanks...I will check the gateway or try other gateway
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
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Hi all,
I've got some problems with the MediaProxy.
I found a UAC with a Public IP that reaches my SIP Server through a NAT. Then the MediaProxy receives the RTP flow from the NAT's IP address, and relay the same flow to the real UAC's IP address (that is public too).
What about the RTP flow sent by the other party? MediaProxy just ignores it.
Anybody knows why is the MediaProxy acting that way?
Thanks in advance,
Víctor
Hi, I am trying to find out whether the following scenario is feasible:
Openser server behind NAT with 5060 forwarded to the box Client 1 behind NAT (different network than the server) - could have ports statically forwarded to it if need be Client 2 on public IP (good news here).
I want Client 1 to be able to register with the server to start with. Also, Client 1 has to be able to call Client 2 (the reverse is not really required but it is probably easier to facilitate) with RTP data flowing directly between he two.
It seems a difficult scenario. I know that it may be possible to use intermediaries (e.g. mediaproxy) but i do not want RTP traffic to go through the server.
Any ideas or pointers to functions / config files.
Regards, Lambros ________________________________________________________________________ Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection.
I use subst to solve this
subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/To:\1$ruri\2/ig')
It works with my OLD gateway
Sam
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
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Sam,
it might work, but it is not correctly from SIP point of view as you change the TO header across the dialog.
regards, bogdan
Sam Hsu wrote:
I use subst to solve this
subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/To:\1$ruri\2/ig')
It works with my OLD gateway
Sam
On 11/28/06, * Bogdan-Andrei Iancu* <bogdan@voice-system.ro mailto: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> > <http://211.111.222.102 >;r2=on;lr=on;ftag=694064c4> > Record-Route: <sip:211.111.222.102 <http://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> > <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> > <http://192.168.123.5:5060>;received= 220.132.138.7 <http://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> > <mailto: sip <mailto:sip>:00928117466@211.111.222.102 <mailto:00928117466@211.111.222.102>>> > From: "joepass" <sip:joepass@211.111.222.102 <mailto:sip:joepass@211.111.222.102> > <mailto:sip <mailto:sip>:joepass@211.111.222.102 <mailto: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> > <mailto: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> <http://votel-tech.com>", uri=" > sip:00928117466@211.111.222.102 <mailto:sip:00928117466@211.111.222.102> > <mailto:sip <mailto:sip>:00928117466@211.111.222.102 <mailto:00928117466@211.111.222.102>>", > response="73a8c869c2a42a12f0d920c2a7d6f068" > P-IPRAuth: votel-tech.com <http://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> > <mailto:sip <mailto:sip>:00928117466@211.111.222.102 <mailto: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 > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list > Users@openser.org <mailto:Users@openser.org> >http://openser.org/cgi-bin/mailman/listinfo/users > >
Bogdan
I understand. Before i get another gateway, i keep on this way. thanks a lot
Sam
On 11/29/06, Bogdan-Andrei Iancu bogdan@voice-system.ro wrote:
Sam,
it might work, but it is not correctly from SIP point of view as you change the TO header across the dialog.
regards, bogdan
Sam Hsu wrote:
I use subst to solve this
subst('/^To:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/To:\1$ruri\2/ig')
It works with my OLD gateway
Sam
On 11/28/06, * Bogdan-Andrei Iancu* <bogdan@voice-system.ro mailto: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. Thestrip()
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 theUAC.
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> > <http://211.111.222.102 >;r2=on;lr=on;ftag=694064c4> > Record-Route: <sip:211.111.222.102 <http://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> > <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> > <http://192.168.123.5:5060>;received= 220.132.138.7 <http://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> > <mailto: sip <mailto:sip>:00928117466@211.111.222.102 <mailto:00928117466@211.111.222.102>>> > From: "joepass" <sip:joepass@211.111.222.102 <mailto:sip:joepass@211.111.222.102> > <mailto:sip <mailto:sip>:joepass@211.111.222.102 <mailto: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> > <mailto: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> <http://votel-tech.com>", uri=" > sip:00928117466@211.111.222.102 <mailto:sip:00928117466@211.111.222.102> > <mailto:sip <mailto:sip>:00928117466@211.111.222.102 <mailto:00928117466@211.111.222.102>>", > response="73a8c869c2a42a12f0d920c2a7d6f068" > P-IPRAuth: votel-tech.com <http://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> > <mailto:sip <mailto:sip>:00928117466@211.111.222.102 <mailto: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 >
> >_______________________________________________ >Users mailing list > Users@openser.org <mailto:Users@openser.org> >http://openser.org/cgi-bin/mailman/listinfo/users > >
-
Bogdan-Andrei Iancu -
lambros21＠netscape.net -
Marnus van Niekerk -
Sam Hsu -
Victor