30 Mar
2005
30 Mar
'05
2:27 p.m.
hi rafael
yes that's it :)
thank you very much
regards
hofi
Hi Hofi
You must use t_check_status... this will solve your problem:
failure_route[1]{
if (t_check_status("408|486")){ ## Do not include 487!!
revert_uri ();
rewritehostport ("x.x.x.x:port");
append_branch();
t_relay();
break;
}
}
regards
Rafael J. Risco G.V.
Millicom Peru SA
On Wed, 30 Mar 2005 11:58:32 +0200 (CEST), hofi <raid(a)hofi.priv.at> wrote:
hi
the first problem is solved (pstn to voicemail !working!)
ser config snip:
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
# ------------- tm parameters
modparam("tm", "fr_timer", 40)
modparam("tm", "fr_inv_timer", 35)
modparam("tm", "wt_timer", 5)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
route {
if (is_user_in("Request-URI", "voicemail")) {
setflag(4);
};
if (lookup("location") | lookup("aliases")) {
if (method == "INVITE" && isflagset(4)) {
log (1, "++++++++++++voicemail***********************");
t_on_failure("1");
avp_write("i:30", "inv_timeout");
}
route(2);
break;
};
}
route [2] {
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
&&
!search("^Route:")){
sl_send_reply("479", "We don't forward to private IP
addresses");
break;
};
if (isflagset(2)) {
force_rtp_proxy();
};
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
}
failure_route[1] {
log (1, "++++++++++++Failure Route******************");
revert_uri();
rewritehost("XXX.XXX.XXX.XXX");
append_branch();
t_relay_to_udp("XXX.XXX.XXX.XXX", "5060");
}
but for my second problem I do not find the error
2) if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs up
ser forwards to asterisk and the voicemail is run in the background, but
instead the call should be cancelled and not forwarded to the voicemail
thx
regards
hofi
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--
rrgv
I never did figure out exactly what the cause was
[sorry], but we were
looking into a secondary carrier and ran into the same problems you
describe below. We took one of our Cisco AS5350's to their NOC to put
in place of their nextone and everything worked just fine. You may
want to look at the settings on your inalp. We were unable to get it
to work with the nextone. My guess was that it had something to do
with codec translations but I have no data to back that up. All the
SIP messaging seemed to be there.
I'd be interested in what fixes this, so if you discover a solution,
please post to the list.
Thanks.
dan
On Tue, 29 Mar 2005 20:36:08 +0200 (CEST), hofi <raid(a)hofi.priv.at>
wrote:
> hi all
>
> configuration:
>
> inalp gw (pstn) ------> ser ------> asterisk (only voicemail)
>
> the following problems occur:
>
> 1) ser forwards the call to asterisk (if the user is in the voicemail
> group) if the client does not answer after 30 sec.
> from sip to sip is working.
> but if a call comes from the pstn to a sip client, the client rings
and
> hangs up after 30 sec. ser then forwards it
to the asterisk, but the
> pstn
> call keeps on ringing and no voicemail is to hear.
>
> ser writes the following error
>
> Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]:
> ++++++++++++INVITE******************
> Mar 29 13:24:39 sip /usr/local/sbin/ser[7017]:
> ++++++++++++voicemail***********************
> Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]:
> ++++++++++++Failure Route******************
> Mar 29 13:25:19 sip /usr/local/sbin/ser[7022]:
> ++++++++++++Failure Route******************
> Mar 29 13:25:19 sip /usr/local/sbin/ser[7017]:
> ++++++++++++voicemail***********************
> Mar 29 13:25:19 sip /usr/local/sbin/ser[7016]: ERROR:
> t_should_relay_response: status rewrite by UAS: stored: 408,
received:
> 200
> Mar 29 13:25:20 sip /usr/local/sbin/ser[7017]: ERROR:
> t_should_relay_response: status rewrite by UAS: stored: 408,
received:
> 200
> Mar 29 13:25:21 sip /usr/local/sbin/ser[7016]: ERROR:
> t_should_relay_response: status rewrite by UAS: stored: 408,
received:
> 200
> Mar 29 13:25:24 sip last message repeated 3 times
> Mar 29 13:25:28 sip /usr/local/sbin/ser[7015]:
> ++++++++++++Loose Route******************
> Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: contact_parser():
Empty
> body Mar 29 13:25:28 sip
/usr/local/sbin/ser[7014]: parse_contact():
> Error
> while parsing
> Mar 29 13:25:28 sip /usr/local/sbin/ser[7014]: get_contact_uri: Error
> while parsing Contact body
>
> 2) if ua1 call ua2 (and ua2 is in the voicemail group) and ua1 hangs
up
> ser forwards to asterisk and the voicemail is
run in the background,
but
> instead the call should be cancelled and not
forwarded to the
voicemail
>
> does anyone know about these problems?
>
> regards
> raid
>
> ser config snip:
>
> loadmodule "/usr/local/lib/ser/modules/sl.so"
> loadmodule "/usr/local/lib/ser/modules/tm.so"
>
> # ------------- tm parameters
> modparam("tm", "fr_timer", 30)
> modparam("tm", "fr_inv_timer", 25)
> modparam("tm", "wt_timer", 5)
> modparam("tm", "fr_inv_timer_avp", "inv_timeout")
>
> route {
> if (lookup("location") | lookup("aliases")) {
> if (is_user_in("Request-URI", "voicemail")) {
> log (1,
"++++++++++++voicemail***********************");
> t_on_failure("1");
> avp_write("i:30", "inv_timeout");
> t_relay();
> route(2);
> break;
> } else {
> log (1, "++++++++++++no
voicemail********************");
> route(2);
> break;
> }
> }
>
> }
>
> route [2] {
> if
(uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
&&
!search("^Route:")){
sl_send_reply("479", "We don't forward to private IP
addresses");
break;
};
if (isflagset(2)) {
force_rtp_proxy();
};
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
}
failure_route[1] {
log (1, "++++++++++++Failure Route******************");
revert_uri();
rewritehostport("XXX.XXX.XXX.XXX:5060");
append_branch();
t_on_failure("1");
t_relay();
break();
}
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