FW: [Serusers] ser and * voicemail - sr-users

1 Oct 2005

I have tethereal on my ser server but didn't see a 408. 

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From: Leon Sun [mailto:leon.sun@keywestcommunications.com] 
Sent: Friday, September 30, 2005 5:41 PM
To: 'Leon Sun'; rthompson(a)vir2com.com
Cc: 'Mark Aiken'
Subject: RE: [Serusers] ser and * voicemail

Hi, guys,

I used Ethereal to trace SIP and found code 408 when it's time out. I
believe we need focus 408 rather than 487.

Sorry Rick, for last e-mail.

Regards

Leon

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From: Leon Sun [mailto:leon.sun@keywestcommunications.com] 
Sent: Friday, September 30, 2005 2:33 PM
To: &#39;rthompson(a)vir2com.com&#39;
Cc: 'Mark Aiken'
Subject: RE: [Serusers] ser and * voicemail

...
 From Mark: 

SER normally sends a 487 when the INVITE timer runs out so you would need to
trigger the voicemail on that event. We have our feature server handing
timeouts not SER (our SER timeout is set to a very large value) so I'm not
sure the best way to proceed. 

Does SER call the failure_route for a locally generated 487 timeout? If so
then rather than the 'break' you have there now just forward to vm. I would
set a different t_on_failure instead of reusing "1" though, so you dont keep
forwarding if the vm fails.

Mark

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From: Rick Thompson [mailto:rthompson@vir2com.com] 
Sent: Friday, September 30, 2005 2:27 PM
To: 'Leon Sun'
Subject: RE: [Serusers] ser and * voicemail

Leon

Thanks for the reply to my post

I see what you are doing here, but that part has been taken care of. When I
get a failure on "INVITE" of  486, 404, 408 or 480 I look in the mysql
database and find the voicemail box for that uri and replace the sip contact
then prefix it with a "V" and relay it to asterisk. Over on the asterisk
side, I route the call to the voicemail context and delete the "V" and send
it to voicemail. All this works except when ser times out on the "INVITE"
and for some reason it doesn't route to asterisk. That's the problem I'm
having.

Rick

  _____  

From: Leon Sun [mailto:leon.sun@keywestcommunications.com] 
Sent: Friday, September 30, 2005 4:36 PM
To: rthompson(a)vir2com.com; serusers(a)lists.iptel.org
Subject: RE: [Serusers] ser and * voicemail

Rick,

I had same problem before and I gave it up since I didn't get any answer
from list. I am using another way(tricky but working) to do voice mail. Hope
it can help you if you can't fix it. 

1.       Check location in your routing parts before relay, if not, forward
to Asterisk.

2.       set up unconditional forward in ATA as 00 + ATA'DID. Make a route
in SER and point 00* to Asterisk.

3.       Strip 00 in Asterisk and send it to voicemail2(${EXTEN})

Regards

Leon Sun

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From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Rick Thompson
Sent: Friday, September 30, 2005 12:58 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] ser and * voicemail

Hi All

I'm working with a ser script, written by someone that's gone now, that
routes inbound calls to an asterisk server for voicemail. The
failure_route[1], sends calls to asterisk and the IVR plays if the ua is
unreachable (not in location) "404", "408" or  the ua is busy
"486" but it
doesn't when the inv time exceeds 30 sec (rings for 30 sec or more). The
call just stops ringing and 10 sec later gets a fast busy. Any ideas from
anyone would be greatly appreciated. Here is the code I'm working with.

failure_route[1] {

            xlog("L_ALERT", "%Tf %mf ****** Failure Route 1: <%rm>
<%rr>
<%rs>\n");

            if(t_check_status("487")) {

                        break;

            };

            if(method=="INVITE" &&
(t_check_status("486|408|404|480"))) {

                        if(avp_db_load("$ruri", "s:mailbox"))

                                    avp_pushto("$ruri/username",
"s:mailbox");

                        prefix("V");

                        rewritehostport("A.B.C.D:5060");

                        append_branch();

                        xlog("L_ALERT", "****** Transfering to
Voicemail\n");

                        t_on_reply("1");

                        t_relay();

            };

}

Thanks

Rick