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PSTN breakout leaves PSTN side connected

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PSTN breakout leaves PSTN side connected

Well this is now the second time that PlusTalk have cut off the SIP side of a call without disconnecting the PSTN side. This is very bad since the recipient of a POTS call cannot hang up the connection, so it ties up their phone line until PlusTalk finally drop the call.

I reported the first case (ticket 20117403) and got a very unsatisfactory response after a few weeks, so I'll see if they do any better with the ticket I just raised for this incident. (ticket 20873133)

Has anyone else been having the same problems?

The trouble is clearly within PlusNet's own systems since the portal logs the time the SIP side of the call was dropped, even though the PSTN side was still up.
3 REPLIES
pacem
Grafter
Posts: 175
Registered: 07-09-2007

PSTN breakout leaves PSTN side connected

Sorry, I don't use PlusTalk; I've read too many issues to believe it is reliable enough to use as a proper phone system.

If it's any concellation. BT will send a hang-up 2 minutes after the recipient of the call puts the phone down. So if it's happening to people you call regularly you could tell them just to leave the phone on the hook for 2 minutes if they find they can't make a call after you have called them.

Paul.<><
glloyd
Rising Star
Posts: 1,652
Thanks: 20
Fixes: 1
Registered: 06-04-2007

PSTN breakout leaves PSTN side connected

What are you using to make calls? I had the same problem when using the Zyzel phone adapter.
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PSTN breakout leaves PSTN side connected

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What are you using to make calls? I had the same problem when using the Zyzel phone adapter.


I have an OpenPBX server talking to PlusNet and a UTStarCom F1000G talking to the OpenPBX server. However, the times in the call log displayed on the PlutNet portal seem to indicate that the problem is actually on the PlusNet side of the connection (i.e. the PlusTalk SIP server knows the call has been terminated but it never tells the server that's actually connected to the PSTN.)

From a technical perspective, I'm not entirely sure how SIP agents are supposed to deal with the network breaking mid-call. Since (as far as I know) there are no SIP messages exchanged during the call and there is no guarantee that the remote agent will always be sending RTP traffic (since agents are allowed to stop sending RTP packets during periods of silence). So once a call has been set up, how is either endpoint to know that the network has broken? I'm afraid my SIP knowledge isn't complete enough.