22 Jan
2008
22 Jan
'08
8:20 p.m.
I have the following scenario:
I receive a request with PSTN_number(a)my.ip.add.res from my VoIP
In alias_db I have defined a rule that points to yyy(a)myip.com which is
a registered account
After alias_db_lookup OpenSER invites as follows:
Invite yyy(a)myip.com
To: xxxxxxx(a)my.ip.add.res
The UA (actually a PBX with VoIP support) answers with 404 not found.
When I invite as follows:
Invite yyy(a)myip.com
To: yyy(a)myip.com
everything works fine.
My question: How do you handle these cases?
Rewriting the To: part seems dificult and will mess up a lot - hein?
Kind regards
Peter
22 Jan
22 Jan
9:41 p.m.
Hi Peter,
According to RFC3261, in SIP, routing is exclusively done based on RURI
(and eventually Route hdrs). TO header has 0 implication in routing.
So, your PBX is deprecated and out of standards and I suggest either
upgrade, either replacement.
Regards,
Bogdan
Peter P GMX wrote:
I have the following scenario:
I receive a request with PSTN_number(a)my.ip.add.res from my VoIP
In alias_db I have defined a rule that points to yyy(a)myip.com which is
a registered account
After alias_db_lookup OpenSER invites as follows:
Invite yyy(a)myip.com
To: xxxxxxx(a)my.ip.add.res
The UA (actually a PBX with VoIP support) answers with 404 not found.
When I invite as follows:
Invite yyy(a)myip.com
To: yyy(a)myip.com
everything works fine.
My question: How do you handle these cases?
Rewriting the To: part seems dificult and will mess up a lot - hein?
Kind regards
Peter
6179
days inactive
6179
days old
1 comments
2 participants
participants (2)
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Bogdan-Andrei Iancu -
Peter P GMX