[Asterisk-Dev] Problem redirecting to voicemail through a SIPproxy (Looks like a bug)

[Chris St Denis]

I'm having a problem redirecting to voicemail. This may be an asterisk bug
I'm not sure, can somebody confirm?

Network layout

GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line.
(Additionally patched with http://bugs.digium.com/view.php?id=2687)
PROXY - Ser version: ser 0.9.3 (i386/freebsd)
FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail.

GATEWAY---PROXY---FEATURE
		|
		|
		UA

For simplicity, hostnames and IPs replaced with the above names. USERNAME,
DSTNUM and SRCNUM also used to replace the UA's username, the source number
of the call, and the destination number of the call. 


The basic SIP dialog goes:

Gateway invites proxy
proxy invites UA
UA replies 180 Ringing.

(Transaction times out and drops to failure route)

PROXY invites FEATURE server (INVITE
sip:*voicemail-busy-2002006 at FEATURE:5060 SIP/2.0.)
PROXY cancels UA
FEATURE replies 200 ok to PROXY
PROXY replies 200 ok to GATEWAY
FEATURE replies 200 ok to PROXY
PROXY replies 200 ok to GATEWAY
UA tells PROXY '487 Request Terminated.'
FEATURE replies 200 ok to PROXY
PROXY replies 200 ok to GATEWAY
...

For some reason, the asterisk gateway doesn't seem to be ACKing the 200 ok.
I don't see any new invite or reinvite going to the gateway which I think
may be what is confusing it.

(I know this is the asterisk users list, but here is a Ser.cfg excerpt for
anyone with experience with both. )

##
# User did not answer phone, or could not connect, or is on the phone and
does not use call waiting.
##
failure_route[3] {
    if(isflagset(10)) {
            if(t_check_status("486")) {
                if (!subst_user('/^/*voicemail-busy-/')){
                        log(1,"Err in subst_user\n");
                }
                xlog("L_ERR", "Relaying to voicemail Busy\n");
            } else {
                if (!subst_user('/^/*voicemail-noanswer-/')){
                        log(1,"Err in subst_user\n");
                }
                xlog("L_ERR", "Relaying to voicemail No answer\n");
            }
            rewritehostport("FEATURE:5060");
            append_branch();
            t_relay();
    }
}

>From the gateway's point of view, the invite looks like 
1. U GATEWAY:5060 -> PROXY:5060
2. INVITE sip:DSTNUM at PROXY SIP/2.0.
3. Via: SIP/2.0/UDP GATEWAY:5060;branch=z9hG4bK6e117757.
4. From: "SRCNUM" <sip:SRCNUM at GATEWAY>;tag=as7f56ca42.
5. To: <sip:DSTNUM at PROXY>.
6. Contact: <sip:SRCNUM at GATEWAY>.
7. Call-ID: 372174c0757805861ab2dcaa54b52a97 at GATEWAY.
8. CSeq: 102 INVITE.
9. User-Agent: Asterisk PBX.
10. Date: Tue, 20 Sep 2005 21:13:57 GMT.
11. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
12. Content-Type: application/sdp.
13. Content-Length: 270.

And the 200 ok looks like 

1. U PROXY:5060 -> GATEWAY:5060
2. SIP/2.0 200 OK.
3. Via: SIP/2.0/UDP GATEWAY:5060;branch=z9hG4bK6e117757.
4. Record-Route: <sip:PROXY;ftag=as7f56ca42;lr>.
5. From: "SRCNUM" <sip:SRCNUM at GATEWAY>;tag=as7f56ca42.
6. To: <sip:DSTNUM at PROXY>;tag=as7f1b7210.
7. Call-ID: 372174c0757805861ab2dcaa54b52a97 at GATEWAY.
8. CSeq: 102 INVITE.
9. User-Agent: Asterisk PBX.
10. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
11. Contact: <sip:*voicemail-busy-USERNAME at FEATURE>.
12. Content-Type: application/sdp.
13. Content-Length: 270.

See links these links for full sip dialog. 

Ngrep from PROXY's point of view	http://pastebin.ca/23469
Ngrep from GATEWAY's point of view	http://pastebin.ca/23470
Ngrep from FEATURE's point of view	http://pastebin.ca/23471