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Unread 03-15-2007, 08:40 PM   #1
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Default Engaged tone v silence


I have a PSTN to SIP provider set-up in my VoXaLot account. When someone calls me and I am busy, my ATA sends 486 Busy Here back to VoXaLot but the person on the PSTN phone does not get an engaged tone, they just get a recorded message telling them that the number does not exist or silence.

If I set-up the same PSTN to SIP provider directly on my ATA instead of through VoXaLot and I am busy, my ATA sends 486 Busy Here back to the provider and the caller gets an engaged tone.

It's as though whatever is being passed back to the provider differs from 486 Busy Here and the provider is interpretting this as an error, rather than busy.

The SIP messages are as follows:

2007-03-15 20:16:04 Local0.Info <010>

SIP/2.0 486 Busy Here<013><010>
To: <sip:*>;tag=d7e4a 13017ad83d6i0<013><010>
From: "+44xxxxxxxxxx" <>;tag=as58e19d59< 013><010>
CSeq: 102 INVITE<013><010>
Via: SIP/2.0/UDP;branch=z9hG4bK0aef.1c13d816.0<013><01 0>
Via: SIP/2.0/UDP;branch=z9hG4bK504bba8c;rport=506 1<013><010>
Record-Route: <sip:;lr=on;ftag=as58e19d59><013><01 0>
Server: Linksys/SPA2102-3.3.6<013><010>Content-Length: 0

2007-03-15 20:16:04 Local0.Info <010>

ACK sip:xxxxxx@ SIP/2.0<013><010>
Via: SIP/2.0/UDP;branch=z9hG4bK0aef.1c13d816.0<013><01 0>
From: "+44xxxxxxxxxx" <>;tag=as58e19d59< 013><010>
To: <sip:*>;tag=d7e4a 13017ad83d6i0<013><010>
CSeq: 102 ACK<013><010>
User-Agent: OpenSer (1.1.0-notls (i386/linux))<013><010>
Content-Length: 0

Any help would be gratefully appreciated. I love the idea of VoXaLot, but I am concerned about my PSTN callers not getting the engaged tone.

Many thanks.

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