Click Here To Visit SIP Broker  

Go Back   Voxalot / SIP Broker Support Forums > Voxalot Forums > Voxalot Support

Voxalot Support Support for the Voxalot service.

 
 
Reply
Thread Tools Display Modes
Unread 03-15-2007, 08:40 PM   #1
738376
Junior Member
 
Join Date: Mar 2007
Posts: 9
Thanks: 1
Thanked 0 Times in 0 Posts
738376 is on a distinguished road
Default Engaged tone v silence

Hi

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 192.168.45.150 <010>

[0]->85.17.19.194:5060(480)
SIP/2.0 486 Busy Here<013><010>
To: <sip:*010xxxxxx@proxy01.eu1.voxalot.com>;tag=d7e4a 13017ad83d6i0<013><010>
From: "+44xxxxxxxxxx" <sip:+44xxxxxxxxxx@eu.voxalot.com>;tag=as58e19d59< 013><010>
Call-ID: 1f72d03661242fec5db07f017dbae47f@eu.voxalot.com<013><010>
CSeq: 102 INVITE<013><010>
Via: SIP/2.0/UDP 85.17.19.194;branch=z9hG4bK0aef.1c13d816.0<013><01 0>
Via: SIP/2.0/UDP 85.17.19.194:5061;branch=z9hG4bK504bba8c;rport=506 1<013><010>
Record-Route: <sip:85.17.19.194;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 192.168.45.150 <010>

[0]<<85.17.19.194:5060(389)
ACK sip:xxxxxx@192.168.45.150:5060 SIP/2.0<013><010>
Via: SIP/2.0/UDP 85.17.19.194;branch=z9hG4bK0aef.1c13d816.0<013><01 0>
From: "+44xxxxxxxxxx" <sip:+44xxxxxxxxxx@eu.voxalot.com>;tag=as58e19d59< 013><010>
Call-ID: 1f72d03661242fec5db07f017dbae47f@eu.voxalot.com<013><010>
To: <sip:*010xxxxxx@proxy01.eu1.voxalot.com>;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.

Chris
738376 is offline   Reply With Quote
Reply

Thread Tools
Display Modes

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Forum Jump

Similar Threads
Thread Thread Starter Forum Replies Last Post
Keep having dial tone while calling maartenvr Voxalot Support 3 05-21-2007 01:34 PM
No busy tone with Betamax via Voxalot tomblandford Voxalot Support 5 04-18-2007 10:03 AM
Sound of silence RDP Voxalot Support 0 03-15-2007 12:56 PM
ATA-186 suddenly no dial tone RDP Voxalot Support 4 12-27-2006 12:55 AM
Weird ringing tone problem pmerrill Voxalot Support 1 06-01-2006 04:15 AM


All times are GMT. The time now is 08:38 AM.


Powered by vBulletin® Version 3.7.2
Copyright ©2000 - 2022, Jelsoft Enterprises Ltd.