asterisk users - Mar 2005 - SER -> Asterisk voicemail on busy/unavailable. Anyone did it? (googling says NO)

Hello all! I googled lists.digium.com and ser mailing list, but did
not find any working configuration of asterisk used as voicemail for
SER. This is my config

        if (uri==myself) {
                if (method=="REGISTER") {
                        save("location");
                        log (1, "Registered\n");
                        break;
                };
                if (lookup("location")) {
                     log (1, "*******  IP to IP call *************");
                     if (method == "INVITE"){
                         setflag (1);
                         t_on_failure("1");
                         t_relay();
                         sl_send_reply ("180", "Ringing");
                        setflag (1);
                         break;
                         }
                     if (!t_relay()) {
                          sl_send_reply("404", "Not Found");
                          break;
                         };

        #        };
        break;
        };


failure_route[1] {
        revert_uri();
        forward(69.70.x.x,5060);
        break();
}

Asterisk sip.conf:

[ser]
host=69.70.x.x
context=ser
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723.1
allow=gsm
allow=ilbc
nat=yes

extensions.conf:

[ser]
include => vm
include => messagecenter

[vm]
exten => _9.,1,VoiceMail(u${EXTEN})
exten => _9.,2,Hangup

[messagecenter]
exten => 555,1,Answer
exten => 555,2,Wait(1)
exten => 555,3,VoiceMailMain(default)
exten => 555,4,Hangup
exten => _555X.,1,Answer                        ; can dial 555<exten>
to skip 'mailbox' prompt.  Useful for speedial.
exten => _555X.,2,Wait(1)
exten => _555X.,3,VoiceMailMain(${EXTEN:3}@default)
exten => _555X.,4,Hangup


All SER calls  9xxx must go to asterisk, and it does, but I get the
following in aster log:
 to 69.70.7.174:5060
Mar  6 18:41:36 WARNING[3539]: chan_sip.c:695 retrans_pkt: Maximum
retries exceeded on call ixiXpRvNGSyIBxmn@192.168.1.103 for seqno 1
(Non-critical Response)
    -- Playing 'beep' (language 'en')
    -- Recording the message
    -- x=0, open writing: 
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: wav49, 0x814cb60
    -- x=1, open writing: 
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: gsm, 0x814d068
    -- x=2, open writing: 
/home/asterisk/var/spool/asterisk/voicemail/default/900/INBOX/msg0005
format: wav, 0x8144980
Mar  6 18:41:45 WARNING[3539]: app.c:619 ast_play_and_record: No audio
available on SIP/69.70.x.x-08149a98??
    -- User hung up
  == Spawn extension (ser, 900, 1) exited non-zero on
'SIP/69.70.x.x-08149a98'
Destroying call 'ixiXpRvNGSyIBxmn@192.168.1.103'


If I use rewritehostport instead of forward, the call does not reach asterisk:

failure_route[1] {
        revert_uri();
        rewritehostport("69.70.x.x:5060");
        t_relay()
        break();

SER log:

4(11513) *******  IP to IP call ************* 1(11506) ERROR:
t_forward_nonack: no branched for fwding
 1(11506) ERROR: w_t_relay (failure mode): forwarding failed
 3(11512) *******  IP to IP call ************* 2(11509) Bye

Is there a way to do append_branch("${EXTEN}@asterisk-box") ?


Anyone did it? Reply pls with your config files!!

>
>
>If I use rewritehostport instead of forward, the call does not reach
asterisk:
>
>failure_route[1] {
>        revert_uri();
>        rewritehostport("69.70.x.x:5060");
>        t_relay()
>        break();
>
>SER log:
>  
>
Your failure route should read:

failure_route[1] {
        revert_uri();
        rewritehostport("69.70.x.x:5060");
        append_branch();   <======YOU MISSED THIS 
        t_relay()
        break();



-- 
Andres
Network Admin
http://www.telesip.net