27 Nov
2006
27 Nov
'06
11:14 p.m.
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(a)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
28 Nov
28 Nov
4 a.m.
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(a)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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Users mailing list
Users(a)openser.org
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4:52 a.m.
Thanks...I will check the gateway or try other gateway
On 11/28/06, Bogdan-Andrei Iancu <bogdan(a)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(a)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(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
8:16 a.m.
New subject: [Users] MediaProxy and Nated Public IP
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
1:09 p.m.
New subject: [Users] Server and 1 client behind NAT
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.
10:23 p.m.
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(a)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(a)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(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
29 Nov
29 Nov
4:32 a.m.
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(a)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(a)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(a)openser.org <mailto:Users@openser.org>
http://openser.org/cgi-bin/mailman/listinfo/users
4:39 a.m.
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(a)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(a)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(a)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(a)openser.org <mailto:Users@openser.org>
http://openser.org/cgi-bin/mailman/listinfo/users
6656
days inactive
6657
days old
8 comments
5 participants
participants (5)
-
Bogdan-Andrei Iancu -
lambros21＠netscape.net -
Marnus van Niekerk -
Sam Hsu -
Victor