Hello,
watch the network traffic with "ngrep -d any -qt port 5060" on your sip
server system and see if the INVITE is sent to asterisk. You can plug
some xlog("__message__") in your configuration to see how the request is
processed and if the INVITE hit the failure_route.
Cheers,
Daniel
On 01/05/07 14:57, raviprakash sunkara wrote:
Hi Users,
I'm Setting UP the Voicemails by integrating with Asterisk and
OpenSER,
After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk,
In openser.cfg ........... is not hiiting the Asterisk server
............. ... any one help me ........
....
....
modparam("tm","fr_timer",6)
modparam("tm","fr_inv_timer",24)
modparam("tm","wt_timer",1)
#mrodparam("tm", "ruri_matching", 0)
#modparam("tm", "via1_matching", 0)
modparam("avpops","avp_url","mysql://root:passwd@192.168.2.75/openser")
modparam("avpops", "avp_table", "usr_preferences")
modparam("avpops","avp_aliases","inv=i:15")
...............
route
{
......
if (loose_route()) {
t_relay();
exit;
};
if(is_method("INVITE") && !has_totag())
{
xdbg("user [$ruri] has voicemail redirection
enabled\n");
# backup R-URI
avp_write("$ruri","$avp(inv)");
setflag(2);
};
..............
....
route(1);
}
route[1] {
if(isflagset(2))
{
t_on_failure("1");
};
}
failure_route[1] {
log("----------------------------------------- \n");
if (t_was_cancelled()) {
xdbg("transaction was cancelled by UAC\n");
return;
}
# restore initial uri
avp_pushto("$ruri", "$avp(inv)");
prefix("9");
# route to Asterisk Media Server
rewritehostport(" 192.168.2.75:5060
<http://192.168.2.75:5060>");
append_branch();
t_relay("192.168.2.75:5060 <http://192.168.2.75:5060>");
resetflag(2);
}
.........................
--
Thanks and Regards
Ravi Prakash Sunkara
ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com>
M:+91 9985077535
O:+91 40 23114549
F:+91 40 40208727
ravi.sunkara(a)hyperion-tech.com <mailto:ravi.sunkara@hyperion-tech.com>
www.hyperion-tech.com <http://www.hyperion-tech.com>
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