Hi Stian,
You can "see" the original TO and FROM uris via $tu and $fu (each can be
broken into $fU (username) and $fd (domain) ). Based on these RO
variables, you can build the new values, like:
uac_replace_from("","sip:$fU@$fd");
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/18/2013 02:07 PM, Stian Øvrevåge wrote:
Thank you.
Using subst_uri I have got rid of the field in the URI.
I'm not sure how to apply the UAC functions.
It is my understanding that they replace the whole To/From fields and
do not return values. The search and replace functions I know of are
in textops and works on the messages. So how do I save the original
To/From, search and replace a part of them, save the result into a new
variable? (And then feed this variable to uac_replace_x).
Brgds,
Stian Øvrevåge
On Sun, Feb 17, 2013 at 9:54 AM, Bogdan-Andrei Iancu
<[email protected]> wrote:
Hi Stian,
The CANCEL is a hop by hop request, which means it is not forwarded by a SIP
proxy, but rather re-generated each time (based on the INVITE is cancels).
So, you cannot change the CANCEL in script, as built based on the INVITE.
Now, the changes you do over the INVITE are not SIP-wise - you cannot simply
change the FROM / TO with text ops, as such changes will not be persistent
during the entire SIP dialog (and as first result, the CANCEL does not
contain the changes you do over the INVITE).
RURI can be changed via textops (see subst_uri() or subst_user() from
textops - http://www.opensips.org/html/docs/modules/1.9.x/textops.html).
For the FROM and TO hdr, use uac_replace_[to|from]() functions from the uac
module - http://www.opensips.org/html/docs/modules/1.9.x/uac.html#id250475 /
Do the changes only over the initial INVITE ! CANCEL and rest will be
automatically changed, no need for script handling for this.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 02/16/2013 04:24 PM, Stian Øvrevåge wrote:
Hi list,
Our upstream SIP provider recently started sending a new field that
makes our UAs ignore most SIP packets. To mediate this I added a
replace_all to remove this field:
replace_all(";phone-context=national", "");
I have moved this to the top of my main route{} block to be as wide as
possible while troubleshooting.
It works great for INVITEs:
Incomming request from upstream:
Request-URI: sip:UserB;[email protected];user=phone
From:<sip:UserA;[email protected];user=phone>;tag=8322
To:<sip:UserB;[email protected];user=phone>
Message routed to UA:
Request-URI: sip:[email protected]:5060
From:<sip:[email protected];user=phone>;tag=8322
To:<sip:[email protected];user=phone>
And all is good.
However, if a CANCEL message is either relayed from upstream or
generated in OpenSIPS, the fields are still present, causing the UA to
not recognize the dialog and continue ringing.
Request-URI: sip:UserB;[email protected];user=phone
From:<sip:UserA;[email protected];user=phone>;tag=8322
To:<sip:UserB;[email protected];user=phone>
Request-URI: sip:[email protected]:5060
From:<sip:UserA;[email protected];user=phone>;tag=8322
To:<sip:UserB;[email protected];user=phone>
I have also added the replace_all to onreply_route[2]{} and
failure_route[1]{}.
In my main block I have a section for CANCEL processing, I started by
adding the replacement there and writing to the log $hdr(From) etc.
But no matter what I did it logged the original values.
How do I fix my CANCEL messages?
And what would be the correct way to log the value of some fields in
an outgoing packet after modification?
Using opensips 1.8.0-notls (x86_64/linux).
Brgds,
Stian Øvrevåge
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