[Serusers] ONSIP Script Modification using avpops for voicemail

Aisling ashling.odriscoll at cit.ie
Mon Sep 26 18:01:28 CEST 2005


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 at gigo.co.uk] 
Sent: 23 September 2005 15:41
To: Aisling
Cc: 'Mark Aiken'; serusers at 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 at gigo.co.uk] 
>Sent: 23 September 2005 10:24
>To: Mark Aiken
>Cc: Aisling; serusers at 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
> >=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
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.


-------------------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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