I have the following call flow:
INVITE -> sip:gonz...@sip.parzee.io -- TLS/TCP/UDP -> *KAMAILIO* - DB
Lookup -> INVITE sip:gonzal...@test.external.com;transport=tls (Phone1)
If Phone1 is Busy or No answer, I want call to go to VM.
Phone1, is not registered to Kamailio, nor I'm using Realtime Integration.
This Phone1 is registered to an external PBX.
Currently in sample configuration script, seems to be that value: $avp(oexten)
is used to redirect to VM, but in my case this value is null.
I didnt find any documentation for this.
*Questions:*
a) What is $avp(oexten) ?
b) What is the best way to pass a Redirect number in SIP INVITE to
VoiceMail system (Asterisk or Freeswitch)
c) Is there a way to configure CFNA timer per alias/uri ?
*Example*:
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
if(!is_method("INVITE|SUBSCRIBE")) return;
# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
xlog("SCRIPT: VoiceMail routing enabled but IP not
defined\n");
return;
}
if(is_method("INVITE")) {
xdbg("SIP Request: method [$rm] from [$fu] to [$tu]\n");
xlog("VoiceMail routing enabled $avp(oexten)\n");
if($avp(oexten)==$null) return;
$ru = "sip:" + $avp(oexten) + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
xlog("SCRIPT: VoiceMail to $tu\n");
} else {
if($rU==$null) return;
$ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
}
route(RELAY);
exit;
#!endif
return;
}
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