Hi Joegen
Where is the SIPHeader header located?
I'm putting the
SIPHeader myHeader( "Remote-Party-Id", "Whatever the value is" );
invite.AddCustomHeader( myHeader );
code in SBCBackDoorTrunk.cxx. inside of the
SBCBackDoorHandler::SBCBackDoorHandler method.. This should call the
constructor first and execute the
SIPHeader myHeader( "Remote-Party-Id", "Whatever the value is" );
invite.AddCustomHeader( myHeader );
Next.
Unless you think it should go somewhere else
Warren Kreckler
Original Message
From: "Joegen E. Baclor"
To: "sales@ER" ;
; "listaopenSBC"
Sent: Wednesday, December 12, 2007 8:46 PM
Subject: Re:
OpenSIPStack B2BUA how to route
sales@ER wrote:
Hi Joegen
I see what you mean. I am not really familiar with the use of the
Remote-Party-Id. We have implemented P-Asserted-Identity for this
instead. Can you point me to the RFC that discusses the use cases for
Remote-Party-Id?
Yes the P-Asserted-Identity replaced the Remote-Party-Id in the RFC but
it
is still in used in older SBC models and my ITSP has not updated the SBC
i
am accessing. I need to modifry the xml code to and and elseif to test
for
this possiblity. Please direct me to the xml that is managing this
identity.
For you be able to rewrite any header before it gets sent to the UAS you
need to override SBCBackDoorCallHandler::OnOutgoingCall();
Look for the declaration of class SBCBackDoorCallHandler in
SBCBackDoorTrunk.cxx. Add a new member function
virtual void OnOutgoingCall(
B2BUAConnection & connection,
B2BUACall & call,
SIPMessage & invite
);
This function will be called whenever there is a new INVITE that will be
sent out by the backdoor trunk. Implement this function right after
BOOL SBCBackDoorCallHandler::OnReceivedMergedInvite() methid in
SBCBackDoorTrunk.cxx
You may add special headers to invite using this code
SIPHeader myHeader( "Remote-Party-Id", "Whatever the value is" );
invite.AddCustomHeader( myHeader );
HTH
Joegen
Warren Kreckler
Original Message
From: "Joegen E. Baclor"
To: "sales@ER"
Sent: Tuesday, December 11, 2007 8:32 PM
Subject: Re:
OpenSIPStack B2BUA how to route
sales@ER wrote:
Hi Joegen
Thank you very much for your replies.
1. I'm using the lastest version.
Then your ITSP must be seeing just a single via. If you think the
contrary, send me a packet capture from sipx->OpenSBC and OpenSBC->ITSP
3. sipX does not re-write header as far as I know. Are you asking
for
sipX header(s) dealing with Caller-ID?
Remote-Party-ID to determine the Calling ID. This is not an element
created
by Sipx. The SBC will need to extract the user part of the From URI
and
create a Remote-Party-ID. I did not see this capability with OpenSBC.
Without this, the called party on the PSTN will either see "Private
Caller"or "Anonymous" on their phone instead of the DID.
I see what you mean. I am not really familiar with the use of the
Remote-Party-Id. We have implemented P-Asserted-Identity for this
instead. Can you point me to the RFC that discusses the use cases for
Remote-Party-Id?
Warren Kreckler
Original Message
From: "Joegen E. Baclor"
To: "sales@ER"
Cc:
Sent: Sunday, December 09, 2007 7:21 PM
Subject: Re:
OpenSIPStack B2BUA how to route
inline...
sales@ER wrote:
Yes They call it peer to peer. By that they meam
1. Via Headers: ITSP has stated that they can accept only 1 Via
statement in an INVITE. As background, each device will add a Via
statement
to the INVITE to if it has processed the INVITE. Only the last or
top
entry
is really of interest to the party that next handles the INVITE. In
order
for ITSP to accept the INVITE of an outbound call, OpenSBC will
need to strip off all previous Via statements from the INVITE and
add
its'
own. I have not found any capability to remove the previously
inserted
Via
What version are you using? There was a bug introduced when we got
back from sipIT 21 due to the changes made there that had the vias
not
getting stripped. Please use the latest CVS. OpenSBC should be
stripping the via before the B2BUA sends the INVITE out to the UAS.
2. Lock IP Address and port to first sender: This option comes into
play
when a call has been answered either by a person or system component
(i.e.
Auto Attendant) and a transfer is attempted. When the transferred
call
is
answered by a new phone or component, it will negotiate use of a new
RTP
port for the media stream. Some service providers, ITSP included,
do not allow the RTP port to change once the initial call is
established.
They do this to protect against the "hijacking" of a call by
Hackers.
Since
the media is flowing through a SBC, the SBC then needs to manage
which
ports
are used to exchange media (voice). If the original port is not
utilized
for the media back to the carrier, the PSTN will not hear any audio
once
the
call is transferred. I do not see this capability with OpenSBC.
In media proxy mode (Always Proxy Media = true), OpenSBC does not
change
the port of RTP even during reInvites.
3. Calling ID: SIPxchange utilizes the From: element to provide
the
Calling ID (DID). It normally inserts the userID in the user part
of
the
From URI. ITSP uses the INVITE element
Remote-Party-ID to determine the Calling ID. This is not an element
created
by Sipx. The SBC will need to extract the user part of the From URI
and
create a Remote-Party-ID. I did not see this capability with
OpenSBC.
Without this, the called party on the PSTN will either see "Private
Caller"or "Anonymous" on their phone instead of the DID.
Can you send a sample of this from header that is rewritten by sipX?
Warren Kreckler
Original Message
From: "Joegen E. Baclor"
To:
Cc:
Sent: Friday, December 07, 2007 12:08 AM
Subject: Re:
OpenSIPStack B2BUA how to route
You need to use the SIP Trunking capability of OpenSBC for this.
Do
you need to authenticate calls with your ITSP?
sales@ER wrote:
Hi
Almost have this puppy working.
Sipx and opensbc generally well understood.
Problem:
When OSBC receives INVITE from sipX => ITSP,
OSBC route the INVITE back to sipX.
We have two rules in the B2Bua route
sip:202.100.2.23:5060 sip: 202.100.2.23 this goes to our ITSP
sip:sipx.sip.net:5060 sip:sipx.sip.net this point to
our
sipx
the missing rule/route?
Where do you put the rule and what should the rule say to route
INVITE
out
to our ITSP?
Warren Kreckler
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