[Serusers] Error forwarding calls to Voicemail from SER
Girish
gr_sh2003 at yahoo.com
Tue Nov 9 11:25:26 CET 2004
Hello,
Just a guess.. Can you add t_check_status (487) in your route[4] and see if the INVITE goes to
Asterisk after cancelling?
--- "Rafael J. Risco G.V." <rafael.risco at gmail.com> wrote:
> Hello
> I have to insist in this issue since I´ve done several test using Sems
> and asterisk with very simple configuration files including the
> original example from ser-cvs... in brief: if I call to a user who
> belongs to voicemail group and I cancel the call before VM forward
> routine begin then an "invite" is sent to a voicemail server
> generating and sending a file with No audio, and I cant account this
> call with "Sip-Response-Code=487" (just an start record without
> stop)...
>
> does someone know how to solve this problem????
>
> thanks in advance
>
> Rafael
>
> PS:
> ser.cfg and asterisk debug for this test:
>
> #
> # SER SIMPLE CFG for VM without acc...
> # ----------- global configuration parameters ------------------------
>
> #debug=3 # debug level (cmd line: -dddddddddd)
> #fork=yes
> #log_stderror=no # (cmd line: -E)
>
> #/* Uncomment these lines to enter debugging mode
> debug=9
> fork=yes
> log_stderror=yes
> #*/
>
> listen=127.0.0.1
> port=5060
>
> # simple proxy script for forwarding to voicemail server
> # for unavailable users
> #
>
> 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/mysql.so"
> loadmodule "/usr/local/lib/ser/modules/group.so"
> loadmodule "/usr/local/lib/ser/modules/usrloc.so"
> loadmodule "/usr/local/lib/ser/modules/registrar.so"
>
> # time to give up on ringing -- global timer, applies to
> # all transactions
> modparam("tm", "fr_inv_timer", 90)
>
> # database with user group membership
> modparam("group", "db_url", "mysql://ser:heslo@localhost/ser")
>
>
> # --------------------- request routing logic -------------------
> route {
>
> if (!mf_process_maxfwd_header("10")) {
> log("LOG: Too many hops\n");
> sl_send_reply("483", "Alas Too Many Hops");
> break;
> };
>
> if (!(method=="REGISTER")) record_route();
> if (loose_route()) {
> t_relay();
> break;
> };
>
> if (!uri==myself) {
> t_relay();
> break;
> };
>
> if (method == "REGISTER") {
> if (!save("location")) {
> sl_reply_error();
> };
> break;
> };
>
> # does the user wish redirection on no availability? (i.e., is he
> # in the voicemail group?) -- determine it now and store it in
> # flag 4, before we rewrite the flag using UsrLoc
> if (is_user_in("Request-URI", "voicemail")) {
> setflag(4);
> };
>
> # native SIP destinations are handled using our USRLOC DB
> if (!lookup("location")) {
> # handle user which was not found
> route(4);
> break;
> };
>
> # if user is on-line and is in voicemail group, enable redirection
> if (method == "INVITE" && isflagset(4)) {
> t_on_failure("1");
> };
> t_relay();
> }
>
> # ------------- handling of unavailable user ------------------
> route[4] {
>
> # non-Voip -- just send "off-line"
> if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) {
> sl_send_reply("404", "Not Found");
> break;
> };
>
> # not voicemail subscriber
> if (!isflagset(4)) {
> sl_send_reply("404", "Not Found and no voicemail turned on");
> break;
> };
>
> # forward to voicemail now
> rewritehostport("200.110.2.132:5060");
> t_relay_to_udp("200.110.2.132", "5060");
> }
>
> # if forwarding downstream did not succeed, try voicemail running
> # at 200.110.2.132:5060
>
> failure_route[1] {
> revert_uri();
> rewritehostport("200.110.2.132:5060");
> append_branch();
> t_relay_to_udp("200.110.2.132", "5060");
> }
>
>
>
> Asterisk Voicemail server sip debug:
> _______________________->>
> *CLI>
>
> Sip read:
> INVITE sip:6609990 at 200.110.2.132:5060 SIP/2.0
> Record-Route: <sip:200.110.2.131;ftag=bb0036aea4;lr=on>
> Via: SIP/2.0/UDP 200.110.2.131;branch=z9hG4bKe24b.b9e800b5.1
> Via: SIP/2.0/UDP 10.0.1.27:5060;rport=5060;branch=z9hG4bKbb0036aea4125
> From: <sip:6603000 at call.millicom.com.pe>;tag=bb0036aea4
> To: <sip:6609990 at call.millicom.com.pe>
> Call-ID: bb9af400-6417-3677-81ae-0002a40055b2 at 10.0.1.27
> CSeq: 125 INVITE
> Supported: timer, replaces
> Min-SE: 1800
> Date: Sun, 05 Jul 1970 12:53:15 GMT
> User-Agent: AddPac SIP Gateway
> Contact: <sip:6603000 at 10.0.1.27:5060>
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
> Content-Type: application/sdp
> Content-Length: 285
> Max-Forwards: 16
> P-hint: usrloc applied
>
=====
Girish Gopinath <gr_sh2003 at yahoo.com>
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