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Unread 12-26-2009, 09:03 PM   #1
jantielens
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Default Outgoing call keeps ringing after disconnect

Hi,

I've got a strange issue over here: I've successfully configured a Voxalot account using two SIP providers and some dialing rules. (I'm planning to use a Voxalot paid account for incoming calls when the service proves to be reliable).

I'm using a Linksys/Cisco PAP2T ATA where the Voxalot account has been configured. I can make outgoing calls without any problems, actually it works great with the dialing rules! :-)

The issue goes as follows: when I make an outgoing call, the called telephone starts ringing (so far so good)? But when I put the phone (that's making the call) back on the hook (disconnect), the called phone keeps ringing (the outgoing call doesn't get disconnected).

I tried a softphone (Nimbuzz) with the same Voxalot account where everything works fine (the outgoing call get's disconnected when I put the calling phone on the hook). So my guess is that it must be a config setting in the PAP2T.

Thanks for your time!
Jan
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Unread 12-26-2009, 09:21 PM   #2
carlosalbffgomes
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Have you followed this tutorial for PAP2 configuration: Setting up Linksys PAP2 - Voxalot FAQ ?
Regards,
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Unread 12-26-2009, 09:23 PM   #3
jantielens
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Yes I did, but it didn't solve the issue ...
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Unread 12-27-2009, 06:04 PM   #4
carlosalbffgomes
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I was thinking about the problem described on this post, and maybe the cause for it relies on NAT (router) and SIP port used by PAP2T. So, my questions are:
  1. Are you using SIP standard port (5060)?
  2. If not (i.e., if you are using some other port such as 80, 443, 2060, 3060 or 4060), are you configuring not only Register Server but also Outbound Proxy Server correctly?
  3. Are you using port 5060 to something else?
  4. Has your router symmetric NAT (bad...!)?
The probable cause for this kind of problems is on SIP port used (maybe not the configured on PAP2T but the one that the NAT on the router assigns to the SIP dialog). Such behaviour explains why callee phone remains ringing when the call is disconnected: it can't send the SIP cancel message to the end point. So, I would check:
  1. STUN Server configuration;
  2. SIP Port configuration (standard or other following the PAP2 tutorial);
  3. Router/NAT behaviour.
Regards,

Last edited by carlosalbffgomes; 12-27-2009 at 06:07 PM.
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