[Serusers] ONSIP Script + Java Rockx ser 0.9.0 script
Iqbal
iqbal at gigo.co.uk
Fri Sep 23 11:23:38 CEST 2005
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
>=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 at cit.ie
> <mailto:ashling.odriscoll at 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 at gigo.co.uk <mailto:iqbal at gigo.co.uk>]
> Sent: 21 September 2005 15:20
> To: Aisling O'Driscoll
> Cc: serusers at lists.iptel.org <mailto:serusers at 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:
>
> >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 script
> >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 of
> >whether they want call forward on no answer or voicemail. They get
> >directed to voicemail if they dial their own extension number....I
> >have posted the modified script which is a combination of the Pauls
> >script and the Onsip script...When I start SER I currently have these
> >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 at onsip.org <mailto:greger at 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 seconds
> >modparam("nathelper", "ping_nated_only", 1)
> >modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock")
> >
> >modparam("cpl-c", "cpl_db", "mysql://root:password@localhost/ser")
> >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", "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
> >=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("$ruri/username", "s:donotdisturb")){
> > # if(avp_check("s:donotdisturb", "eq/y/i")){
> > # route(x) #whereever asterisk voicemail is
> > # break;
> > # };
> > #};
> >
> > #
> -----------------------------------------------------------------
> > # Record Route Section
> > #
> -----------------------------------------------------------------
> > if (method!="REGISTER") {
> > record_route();
> > };
> >
> > if (method=="BYE" || method=="CANCEL") {
> > unforce_rtp_proxy();
> > };
> >
> > if (method=="INVITE"){
> > if(is_user_in("Request-URI", "voicemail")){
> > setflag(31);
> > };
> > };
> >
> > #
> >----------------------------------------------------------------------
> >--
> > # URI Compare Section
> > #
> > # Here we compare the "from" and "to" to see if the
> caller is
> >dialing
> > # their own extension. If so then we route to voicemail if
> >needed
> > #
> >----------------------------------------------------------------------
> >--
> > if (method=="INVITE") {
> > avp_write("$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 failed");
> > };
> > 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 sl_send_reply()
> 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, caller==callee)
> >
> > 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", "$voicemail")){
> > avp_delete("$voicemail");
> > route(9);
> > break;
> > };
> >}
> >
> >
> >
> >-------------------Legal
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> >
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> >
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> -------------------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.
>
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