On 10.08.2009 17:21 Uhr, karhu wrote:
I will have to generate an INVITE later on so i am
using this topic to ask
you something. I'm working on having in one side a sequence (BYE +
(re-INVITE+ newSDP)). The new SDP will have some more media parameters than
the initial one.
I have been reading documentation, and it seems there is an issue to
generate a bye, by using :
dlg_bye(side) function from dialog module or uac_req_send() from uac module.
I've been trying to use these functions in Kamailio.cfg but it doesn't
generate any bye at all on wireshark trace !!
Using dialog module :
Informations about caller, callee, callid are well stored into the DB but
When i am calling :
dlg_bye("callee") no BYE is send to the callee...
note that this function works only for established dialogs -- so you
have to track the call with dialog module.
Using uac module :
$uac_req(method)="BYE";
$uac_req(furi)="sip:bob@10.10.10.2"
$uac_req(turi)="sip:alice@10.10.10.2"
uac_req_send();
to execute one of these functions i puted them in route[66] with a xlog line
and using rtimer module to be sure it is well executed. xlog line is
executed but no BYE on wireshark :s
Probably the filter is not right. Watch all the network interfaces you have.
About the INVITE+SDP :
Is it possible to generate an INVITE from the cfg file ? will uac_req_send()
can have a body on his method? i guess no !!
You can have a body with the uac_req_send(), see the readme of the uac
module. But probably won't help you much, since kamailio does not handle
media.
Cheers,
Daniel
The only way i found to generate INVITE+SDP is to use
MI commands from tm
module : t_uac_dlg !! am i right ?
Best regards,
karhu
--
Daniel-Constantin Mierla
* SIP Router Bootcamp
* Kamailio (OpenSER) and Asterisk Training
* Berlin, Germany, Sep 1-4, 2009
*
http://www.asipto.com/index.php/sip-router-bootcamp/