Re: [reSIProcate-users] Can't find session when received a refer call with replace header.

[Herbert Karajan <boost.regex@xxxxxxxxx>]

So this is the server's issue ? I have did more test, it's seems the opensips and iptel replace the call-id when it relay the INVITE.

For example, A call B, the call-id is 1234 in the outgoing INVITE from A to server  opensips/iptel, but when the B received INVITE from server opensips/iptel, the call-id is not 1234, it's another value.

Best regards,

On Tue, May 13, 2014 at 9:02 PM, Scott Godin <sgodin@xxxxxxxxxxxxxxx> wrote:

In the working flow, from the first 200 response you can see the dialog info is:

Call-ID: BxY5GHi7NPgZo0H8WRrCiA..

to-tag: cf38b430

from-tag: 1f346342

The resulting Replaces header in the INVITE to the 3rd party is correct:

Replaces: BxY5GHi7NPgZo0H8WRrCiA..;to-tag=cf38b430;from-tag=1f346342

In the non-working example you see:

Call-ID: 118441ED-53719E6D0004983B-84A43700

to-tag: 8336df3f

from-tag: 1CE9012D-53719E6D0004982A-84A43700

In this case the resulting Replaces header is wrong from User-Agent: FLASH APP:

Replaces: T6kkMa3KyPWHNzvCewUBfw..;to-tag=513B7FE8-53719E6D000490F1-3492C700;from-tag=df24e461

Regards,

Scottt

On Tue, May 13, 2014 at 12:36 AM, Herbert Karajan <boost.regex@xxxxxxxxx> wrote:

Hi Scott, thanks for attention, attached the wireshark traces in email, the test flows are:

1. jkweb1 call jkweb3

2. jkweb1 hold the call to jkweb3

3. jkweb1 call jkweb2

4. jkweb1 execute the attend transfer to jkweb2 to replace the jkweb3

5. The jkweb3 received the incoming call from jkweb2 with replace header

6. iptel.org: the "mDum.findInviteSession" fails;

    sip2sip: works fine.

Best regards,

On Tue, May 13, 2014 at 3:28 AM, Scott Godin <sgodin@xxxxxxxxxxxxxxx> wrote:

Perhaps sip.iptel.org has the to and from tags backwards in the embedded dialog info in the replaces header - impossible to tell without wireshark traces or logs.

Regards,

Scott

On Sun, May 11, 2014 at 11:25 AM, Herbert Karajan <boost.regex@xxxxxxxxx> wrote:

Hi, I'm test my UA with sip.iptel.org and opensips.org for attended transfer feature.

I'm follow the code that in the basicclientcall.cxx:

=================

void

BasicClientCall::onNewSession(ServerInviteSessionHandle h, InviteSession::OfferAnswerType oat, const SipMessage& msg)

{

   InfoLog(<< "onNewSession(ServerInviteSessionHandle):  msg=" << msg.brief());

   mInviteSessionHandle = h->getSessionHandle();

   // First check if this INVITE is to replace an existing session

   if(msg.exists(h_Replaces))

   {

      pair<InviteSessionHandle, int> presult;

      presult = mDum.findInviteSession(msg.header(h_Replaces));

      if(!(presult.first == InviteSessionHandle::NotValid())) 

      {         

===================

Here always fail to  mDum.findInviteSession, the presult.first is equals to InviteSessionHandle::NotValid().

But when I test with sip2sip.info it works fine.

Best regards,

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