26 Sep
2005
26 Sep
'05
12:01 p.m.
Hello,
Following on from what I said below, I changed my ser.cfg (bearing in
mind that my ser.cfg is the onsip call feature on with rtpproxy
support)to support asterisk voicemail according to my plan.....At first
I am only dealing with leaving the voicemail.
In route[3] (The INVITE handler route), I added:
if(avp_db_load("$ruri/username", "s:voicemail")){
if(avp_check("s:voicemail", "eq/y/I")){
setflag(5);
};
};
Then in failure route I changed the end to:
if(isflagset(27) && t_check_status("408")){
if(avp_pushto("$ruri", "s:fwdnoanswer")){
avp_delete("s:fwdnoanswer");
resetflag(27);
route(6);
break;
};
}
else{
if(isflagset(5) && t_check_status("408")){
resetflag(5);
log(1, "in failure route - flag 5 is set, should be
routing to route 7");
break;
};
};
unforce_rtp_proxy();
}
route[7]{
log(1, "In route 7!! - before revert uri");
revert_uri();
log(1, "In route 7!! - before rewritehostport");
rewritehostport("x.x.x.x:5064");
log(1, "In route 7!! - before append branch");
append_branch();
log(1, "In route 7!! - before t relay to udp");
t_relay_to_udp("x.x.x.x", "5064");
log(1, "In route 7!! - before break");
break;
}
Now what's strange is that all the log messages are printed meaning the
flag is being set (because voicemail is set to 'y' in the
usr_preferences table), the failure route is being entered and its going
to route 7, but strangely enough a 404 Not Found message is being sent
to the phone....Does anyone have any ideas???
Many thanks as always for help.
-----Original Message-----
From: Iqbal [mailto:iqbal@gigo.co.uk]
Sent: 23 September 2005 15:41
To: Aisling
Cc: 'Mark Aiken'; serusers(a)lists.iptel.org
Subject: Re: [Serusers] ONSIP Script + Java Rockx ser 0.9.0 script
seems fine I have done it in grp table, and use is_user_in, "voicemail"
check instead and set a flag, I am not sure whats best practive, and
sure usr_preferences is the way to go, but I decided to use that for
features like callfwd, donotdisturb, as opposed to voicemail which I
figured fell into a diff category. Either way it should be fine.
Once the check is done, flags set the rest is fine.
Now for playing voicemail, what i do is near the top of my INVITE
handler i see what number has been dialied, so if the user dials 699 (my
voicemail number for all users), it sends straight to asterik, where
they get prompted for username/pass combo to listen to voicemail, and
then to use externally what I did was to bind 123456789 to my ser, and
again if that gets dialed send to asterisk to pickup users voicemail
also.
Iqbal
Aisling wrote:
Hi Iqbal,
Thanks for the detailed reply....I have an outline plan for how my
asterisk/ser setup will work (using the onsip callforward script with
rtp proxy)....If the logic is messed up please let me know because I'm
trying to minimize the chance of making stupid changes to the config
which will affect other things....Then hopefully if this makes sense it
might help other people....
#########################Leaving
voicemail################################
1) In the usr_preferences table in the ser database have an entry for
user 2092.
Insert into usr_preferences (username, attribute, value) values
("2092",
"voicemail", "y");
2) In Route[3] (used for call invite handling)
if(avp_db_load("$ruri/username","s:voicemail")){
if(avp_check("s:voicemail", "eq/y/I")){
setflag(5);
};
};
This will check if the user wants to use voicemail according to the
preference that is set for them in the usr_preferences table. I they
don't want to use voicemail set value to "n"
3) In failure route[1]
if(isflagset(5) && t_check_status("408")){
avp_delete("s:voicemail");
resetflag(5);
append_branch();
route(x);
break;
};
4) route[x]
{
acc_db_request("missed called", "missed_calls");
revert_uri();
rewritehostport("x.x.x.x:5064");
append_branch();
t_relay_to_udp(x.x.x.x", "5064");
break();
}
###################################Playing
Voicemail#######################
1) as above
2) as above
3) If the user dials their own username they will be diverted to
asterisk
if (method=="INVITE"){
if(avp_check("i:34", "eq/$ruri/i")){
if(isflagset(5)){
route(x);
break;
} else{
sl_send_reply("486", "Busy");
break;
};
};
};
4) route[x] as before
Aisling.
-----Original Message-----
From: Iqbal [mailto:iqbal@gigo.co.uk]
Sent: 23 September 2005 10:24
To: Mark Aiken
Cc: Aisling; serusers(a)lists.iptel.org
Subject: Re: [Serusers] ONSIP Script + Java Rockx ser 0.9.0 script
Hi
@Mark: I agree, but some people only really need ser, or should I say
if
your route to much to asterisk its pointless having ser, I keep my
Ip__IP traffic in ser, in fact my billing there also, asterisk as you
mentioned for features, and when my upstream gateway wont support REFER
to do transfers, and cool things like call pickup.
@Aisling:
if (method=="INVITE"){
if(is_user_in("Request-URI", "voicemail")){
setflag(31);
};
};
what this will do is to see if the username you have in RURI is in your
grp table and has an entry of voicemail next to it, if
so it will set a
flag 31
if (method=="INVITE") {
avp_write("$from",
"i:34");
if (avp_check("i:34", "eq/$ruri/i")) {
if (isflagset(31)) {
route(8);
break;
} else {
sl_send_reply("486", "Busy");
break;
};
};
};
what this does is basically to see if your are calling yourself, i.e if the from and the RURI is the same, first it stores the
from setting,
and
the compares that to your ruri.Now if they are the same, it will see if
flag 31 is set, i.e if you have voicemail attached to
the RURI in grp
table. If both are true then goto route8, at which point it seems as if
your are relaying to another server.
You seem to be getting busy, which means that flag 31 is not being set,
check your ngrep dump see what your RURI has, and see
what entries your
grp table has.
The problem with matching like this (or at least the problem(s) I had)
was that there are alot of different call scenarios which all need to
be
taken into account.
EG my usernames are not e164 numbers, they are part of that number,
which means my ruri, and from can be different to the e164 format.
Now when a caller calls from 040600, to 0845040602, it should see if
0845xxx is busy, send to voicemail, but in grp ther is no 0845xxx,
hence
I need to make sure that the check/match is done after the alias
lookup.
I had to move lookup(aliases) near the to to make sure all my lookups
had been done at the start, and then I can start matching.
Your best friend will be ngrep, if you start doing the diff call
scenarios, and seeing what needs to be matched, you can set 1 run
settings, once it all works work out a tidier logic.
If you throw asterisk into the loop, you add another headache
(depending
on what you do), cause asterisk was really designed for ip devices to
register with it, but if you register in ser, you could lose some of
that info (although latest beta, has some nice SIP header parsing, to
pull info from).
The other problem you may hit with voicemail, depends on how you set
your username, initially my subscriber table had names for
username...eg
Iqbal, and the alias would be 040600, or whatever, problem with this
was
that voicemail in asterisk is better with number, especially when you
have to dial the mailbox number from your phone :-), so I switched the
subscriber table to hold the number, cause I think unless you will be
using software dialers, no one will type in names.
"voicemail=1:500;calltype=i:700;fwd_no_answer_type=1:701;fwd_busy_type
of
get
number....I
Pauls
these
seconds
"mysql://root:password@localhost/ser")
"mysql://root:password@localhost/ser")
> >modparam("avpops", "avp_table",
"usr_preferences")
> >modparam("avpops", "avp_aliases",
>
>"voicemail=1:500;calltype=i:700;fwd_no_answer_type=1:701;fwd_busy_type
> #------------------------------------------------------------------
#------------------------------------------------------------------
#------------------------------------------------------------------
#------------------------------------------------------------------
#------------------------------------------------------------------
is
if
failed");
#------------------------------------------------------------------
sl_send_reply()
caller==callee)
"$voicemail")){
-------------------Legal
Disclaimer---------------------------------------
=1:702")
All this does is to define aliases, you can use it, or not
iqbal
Mark Aiken wrote:
I see a lot of people trying to use SER and
Asterisk together but
mixing the roles in strange and complex ways.
Asterisk is an excellent SIP "feature server" and "b2bua" where SER
is
an excellent SIP proxy, registrar, and far-end NAT
solution.
If you design your network based on those strengths you can create a
solution that rivals 7 figure commercial solutions. People that do
otherwise seem to just fall over themselves.
Mark
On 9/21/05, *Aisling* <ashling.odriscoll(a)cit.ie
<mailto:ashling.odriscoll@cit.ie>> wrote:
Many thanks Iqbal- That was the problem.
I think the issue is that I don't even really understand what that
line
does or how the code that I took from Paul's config affected my
config
file.....SER now displays no errors but is
still not working as
expected. For example I expect that if a phone (e.g. 2092) dials
its own
extension, it should be caught by
if (method=="INVITE"){
if(is_user_in("Request-URI", "voicemail")){
setflag(31);
};
};
This then should be caught by the URI Compare code below and
routed to
route[8]....
However that's not happening and a 486 Busy is being returned to
the
phone. I find this hard to troubleshoot
because I think the code
should
be correct...given my current understanding anyhow...
If Iqbal (or anyone else) has a clearer understanding of this and
can
shed some light on it, that would be great.
Kindest Regards,
Aisling.
-----Original Message-----
From: Iqbal [mailto:iqbal@gigo.co.uk <mailto:iqbal@gigo.co.uk>]
Sent: 21 September 2005 15:20
To: Aisling O'Driscoll
Cc: serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org>
Subject: Re: [Serusers] ONSIP Script + Java Rockx ser 0.9.0 script
"voicemail=1:500;calltype=i:700;fwd_no_answer_type=1:701;fwd_busy_type
=1:702")
try a voicemail=i:500 not 1:500
Iqbal
Aisling O'Driscoll wrote:
script
Hello everyone,
I am trying to modify the Onsip call forward script to forward to
asterisk voicemail given a users preference. I am using the
that Java
RockX (Paul) posted in January...
http://lists.iptel.org/pipermail/serusers/2005-January/014968.html
I think I am getting very confused though with the logic...I want
users to be able to load their preference using the avpops module
whether
they want call forward on no answer or voicemail. They
directed
to voicemail if they dial their own extension
have
posted the modified script which is a combination of the
script and
the Onsip script...When I start SER I currently have
errors but
I'd say these are a consequence of my messed up logic
;)....
0(20031) ERROR:parse_avp_name: unsupported type [1]
0(20031) ERROR:add_avp_galias_str: <500> not a valid AVP name
0(20031) parse error (70,20-21): Can't set module parameter
The config is below....
If anyone can offer some insight, Id be very grateful.
Many thanks,
Aisling.
# $Id: nat-rtpproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V.
Teigre/emailAddress= greger(a)onsip.org <mailto:greger@onsip.org> $
#debug=3
#fork=yes
#log_stderror=no
#listen=x.x.x.x # INSERT YOUR IP ADDRESS HERE
#port=5060
#children=4
check_via=no
dns=no
rev_dns=no
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://root:password@localhost/ser"
alias="x.x.x.x:5060"
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/cpl-c.so"
loadmodule "/usr/local/lib/ser/modules/avpops.so"
loadmodule "/usr/local/lib/ser/modules/permissions.so"
loadmodule "/usr/local/lib/ser/modules/speeddial.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
modparam("auth_db|permissions|uri_db|usrloc", "db_url",
"mysql://root:password@localhost/ser")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "password_column", "password")
modparam("nathelper", "natping_interval", 30) #ping every 30
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock")
modparam("cpl-c", "cpl_db",
modparam("cpl-c", "cpl_table", "cpl")
modparam("cpl-c", "cpl_dtd_file",
"/tmp/ser-0.9.3/modules/cpl-c/cpl-06.dtd")
modparam("cpl-c", "lookup_domain", "location")
modparam("tm", "fr_inv_timer", 27)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
modparam("usrloc", "db_mode", 2)
modparam("registrar", "nat_flag", 6)
modparam("rr", "enable_full_lr", 1)
modparam("speeddial", "db_url",
"mysql://root:password@localhost/ser")
modparam("speeddial",
"user_column", "userid")
modparam("speeddial", "sd_user_column", "short_user")
modparam("speeddial", "sd_domain_column", "short_domain")
modparam("speeddial", "new_uri_column", "real_uri")
modparam("avpops", "avp_url",
=1:702")
> >
> >route {
> >
> > #
> -----------------------------------------------------------------
> > # Sanity Check Section
> > #
> -----------------------------------------------------------------
> > if (!mf_process_maxfwd_header("10")) {
> > sl_send_reply("483", "Too Many Hops");
> > break;
> > };
> >
> > if (msg:len > max_len) {
> > sl_send_reply("513", "Message Overflow");
> > break;
> > };
> >
> >
>
>
>
#------------------------------------------------------------------
#
NOTIFY Keep Alive Section
if
((method=="NOTIFY") && search("^Event: keep-alive")){
sl_send_reply("200", "OK");
break;
};
#
Speed Dialing Section
if
((method=="INVITE") && (uri=~"^sip:[0-9]{2}@.*")) {
sd_lookup("speed_dial");
};
#
Do not Disturb Section
#if(avp_db_load("$ruri/username", "s:donotdisturb")){
# if(avp_check("s:donotdisturb", "eq/y/i")){
# route(x) #whereever asterisk voicemail
#
break;
# };
#};
#
-----------------------------------------------------------------
# Record Route Section
#
-----------------------------------------------------------------
if (method!="REGISTER") {
record_route();
};
if (method=="BYE" || method=="CANCEL") {
unforce_rtp_proxy();
};
if (method=="INVITE"){
if(is_user_in("Request-URI", "voicemail")){
setflag(31);
};
};
#
----------------------------------------------------------------------
--
# URI Compare Section
#
# Here we compare the "from" and "to" to see if the
caller is
dialing
# their own extension. If so then we route to voicemail
needed
#
----------------------------------------------------------------------
--
if (method=="INVITE") {
avp_write("$from", "i:34");
if (avp_check("i:34", "eq/$ruri/i")) {
if (isflagset(31)) {
route(8);
break;
} else {
sl_send_reply("486", "Busy");
break;
};
};
};
#
-----------------------------------------------------------------
# Loose Route Section
#
-----------------------------------------------------------------
if (loose_route()) {
if (has_totag() && (method=="INVITE" ||
method=="ACK"))
{
if
(nat_uac_test("19")) {
setflag(6);
force_rport();
fix_nated_contact();
};
force_rtp_proxy("l");
};
route(1);
break;
};
#
-----------------------------------------------------------------
# Call Type Processing Section
#
-----------------------------------------------------------------
if (uri!=myself) {
route(4);
route(1);
break;
};
if (method=="CANCEL") {
route(1);
break;
} else if (method=="INVITE") {
if(!cpl_run_script("incoming", "is_stateless"))
{
# script execution failed
t_reply("500", "CPL script execution
};
route(3);
break;
} else if (method=="REGISTER") {
#handle REGISTER messages with CPL script
cpl_process_register();
route(2);
break;
};
lookup("aliases");
if (uri!=myself) {
route(4);
route(1);
break;
};
if (!lookup("location")) {
if(isflagset(31)){
setflag(19);
};
#sl_send_reply("404", "User Not Found");
#break;
};
route(1);
}
route[1] {
#
-----------------------------------------------------------------
# Default Message Handler
#
-----------------------------------------------------------------
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" && isflagset(6)) {
unforce_rtp_proxy();
};
sl_reply_error();
};
}
route[2] {
#
-----------------------------------------------------------------
# REGISTER Message Handler
#
----------------------------------------------------------------
if (!search("^Contact:[ ]*\*") && nat_uac_test("19"))
{
setflag(6);
fix_nated_register();
force_rport();
};
sl_send_reply("100", "Trying");
if (!www_authorize("","subscriber")) {
www_challenge("","0");
break;
};
if (!check_to()) {
sl_send_reply("401", "Unauthorized");
break;
};
consume_credentials();
if (!save("location")) {
sl_reply_error();
};
}
route[3] {
#
-----------------------------------------------------------------
# INVITE Message Handler
#
-----------------------------------------------------------------
if (!proxy_authorize("","subscriber")) {
proxy_challenge("","0");
break;
} else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
#consume_credentials();
if (nat_uac_test("19")) {
setflag(6);
}
lookup("aliases");
if (uri!=myself) {
route(4);
route(1);
break;
};
#Blind Call Forwarding
if (avp_db_load("$ruri/username", "s:callfwd")){
setflag(22);
avp_pushto("$ruri", "s:callfwd");
route(6);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
if (avp_db_load("$ruri/username", "s:fwdbusy")){
if(!avp_check("s:fwdbusy", "eq/$ruri/i")){
setflag(26);
};
};
if (avp_db_load("$ruri/username", "s:fwdnoanswer")){
if(!avp_check("s:fwdnoanswer", "eq/$ruri/i")){
setflag(27);
};
};
t_on_failure("1");
route(4);
route(1);
}
route[4] {
#
-----------------------------------------------------------------
# NAT Traversal Section
#
-----------------------------------------------------------------
if (isflagset(6)) {
force_rport();
fix_nated_contact();
force_rtp_proxy();
};
}
route[5]{
#-----------------------------------------------------------------
# PSTN Handler
#-----------------------------------------------------------------
revert_uri();
rewritehostport(" x.x.x.x:5064");
append_branch();
t_relay_to_udp("x.x.x.x", "5064");
break();
t_on_failure("1");
route(4);
route(1);
}
route[6]{
#
Call Forwarding Handler
#
# This must be done as a route block because
cannot
#-------------------------------------------------------------------
# be called from the failure_route block
if(uri=~"^sip:1[0-9][10]@"){
strip(1);
};
lookup ("aliases");
if(uri!=myself){
if(!isflagset(22)){
append_branch();
};
route(4);
route(1);
break;
};
if(uri=~"^sip:011[0-9]*@"){
route(4);
route(5);
break;
};
if(!lookup("location")){
if (uri=~"^sip:[0-9][10]@"){
route(4);
route(1);
break;
};
sl_send_reply("404", "User Not Found");
};
route(4);
route(1);
}
route[8] {
# voicemail route #2
#
# this path this executed during these conditions:
#
# cond 1) the called number is not in the location
table
# cond 2) the from_uri == to_uri
(ie,
if (method == "INVITE" || method == "ACK"){
force_rtp_proxy();
};
rewritehostport("x.x.x.x:5064");
t_on_reply("1");
if (!t_relay()) {
if(method == "INVITE" || method == "ACK"){
unforce_rtp_proxy();
};
sl_reply_error();
};
}
route[9]{
#voicemail route 1
#
#this path is executed during these conditions:
#
# cond 1) the called number is in the location table
# but the callee did not answer the phone ( i.e.
failover
to
voicemail)
rewritehostport("x.x.x.x:5064");
append_branch();
t_on_reply("1");
if (!t_relay()){
if(method == "INVITE" || method=="ACK"){
unforce_rtp_proxy();
};
sl_reply_error();
};
}
onreply_route[1] {
if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {
if (!search("^Content-Length:[ ]*0")) {
force_rtp_proxy();
};
};
if (nat_uac_test("1")) {
fix_nated_contact();
};
}
failure_route[1]{
if(t_check_status("487")){
break;
};
if(isflagset(26) && t_check_status("486")){
if(avp_pushto("$ruri", "s:fwdbusy")){
avp_delete("s:fwdbusy");
resetflag(26);
append_branch();
route(6);
break;
};
};
#Here we can have either voicemail OR forward no answer
#forward on no answer is flag 27
#voicemail is flag 31
if(isflagset(27) && t_check_status("408")){
if(avp_pushto("$ruri", "s:fwdnoanswer")){
avp_delete("s:fwdnoanswer");
resetflag(27);
append_branch;
route(6);
break;
};
}
else if(isflagset(31) && avp_pushto("$ruri",
avp_delete("$voicemail");
route(9);
break;
};
}
The above electronic mail transmission is confidential and intended
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person to whom it is addressed. Its contents may be
protected by legal and/or professional privilege. Should it be received
by you in error please contact the sender at the above quoted email
address. Any unauthorised form of reproduction of this message is
strictly prohibited. The Institute does not guarantee the security of
any information electronically transmitted and is not liable if the
information contained in this communication is not a proper and complete
record of the message as transmitted by the sender nor for any delay in
its receipt.
.
-------------------Legal
Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only
for the person to whom it is addressed. Its contents may be protected by
legal and/or professional privilege. Should it be received by you in
error please contact the sender at the above quoted email address. Any
unauthorised form of reproduction of this message is strictly
prohibited. The Institute does not guarantee the security of any
information electronically transmitted and is not liable if the
information contained in this communication is not a proper and complete
record of the message as transmitted by the sender nor for any delay in
its receipt.
-------------------Legal Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only for the person to
whom it is addressed. Its contents may be protected by legal and/or professional
privilege. Should it be received by you in error please contact the sender at the above
quoted email address. Any unauthorised form of reproduction of this message is strictly
prohibited. The Institute does not guarantee the security of any information
electronically transmitted and is not liable if the information contained in this
communication is not a proper and complete record of the message as transmitted by the
sender nor for any delay in its receipt.