[Serusers] SER with VoiceMail Problem

[Tracy Lofton]

Hi Ram,
I do not understand your setup.  You are on static IP?  Openser/Asterisk on 
same machine or different machines?

When you dial ext. 8888, it does transfer to Asterisk voicemail?  Does the 
phone ring on the phone first ("no user") before Asterisk's IVR starts?

Do you get to the PSTN gw and then Asterisk IVR picks up and bypasses the 
Dial() application for extension 8888 in Asterisk?  Is you PSTN gw via 
Asterisk or voip provider?

Have you setup your user info in Asterisk appropriately?

I am trying to get a clearer visual.

Tracy

>From: ram <talk2ram at gmail.com>
>To: serusers at iptel.org
>Subject: [Serusers] SER with VoiceMail Problem
>Date: Mon, 28 Aug 2006 18:45:10 +0530
>
>Hi
>
>i am new to ser,
>iam going through the  list,
>i found some of the similar queries
>
>http://lists.iptel.org/pipermail/serusers/2003-October/003132.html
>
>I have used the config file
>changed according to my setup
>and iam able to pick up messages from Asterisk also
>
>But i have some problem here is
>
>1. i have registered user 8888 in my SER, when i dial 8888
>its going to voice mail, how can i set the caller==calee Busy
>
>even i have added this
>        if (!uri==myself) {
>                t_relay();
>                break;
>        };
>
>But still no use.
>
>2. when i dial to PSTN number, when the PSTN also go to voice mail after
>certain rings
>I suppose to get the voice Menu of other Side Number, but the voice go 
>Blank
>( instead of going to VoiceMenu)
>
>when checked in my Log, i get this error in my syslog
>
>Aug 28 18:33:55 router ser[757]: no location or no user
>Aug 28 18:34:01 router ser[751]: 10 digit exp match w/leading 1
>Aug 28 18:34:01 router ser[751]: route[2]:SIP-to-PSTN call routed
>Aug 28 18:34:10 router ser[757]: no location or no user
>Aug 28 18:34:25 router ser[758]: no location or no user
>Aug 28 18:34:28 router ser[791]: ACC: call missed: from=88888 <
>sip:88888 at mydomain.com>;tag=1674409670, i-uri= 
>sip:1xxxxx80369 at mydomain.com,
>method=INVITE, o-uri=sip:1xxxxx80369 at myprovider:5060, code=408 Request
>Timeout
>Aug 28 18:34:28 router ser[791]: failure_route[5]
>Aug 28 18:34:28 router ser[791]: route[5]: re-relay to PSTN
>Aug 28 18:34:28 router ser[751]: ERROR: t_should_relay: status rewrite by
>UAS: stored: 408, received: 487
>Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1
>Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed
>Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1
>Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed
>Aug 28 18:34:40 router ser[751]: no location or no user
>
>
>any suggestions
>
>Ram


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