Re: [Serusers] Error forwarding calls to Voicemail from SER

hello Girish I did some test using t_check_status ("487") in failure_route[1] like this: failure_route[1] { if (t_check_status("487")){ revert_uri (); rewritehostport ("200.110.2.132:5060"); append_branch(); t_relay_to_udp("200.110.2.132", "5060"); } ...but did not work, can you send us more details about how to use t_check_status("487") in route[4]? thanks rafael On Tue, 9 Nov 2004 02:25:26 -0800 (PST), Girish <gr_sh2003(a)yahoo.com> wrote: > 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(a)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"); > > } > > > >