Help with fxo tried all solution but stil cant hangup

currently i setup an asterisk server. with tdm400 card (2fxo & 2fxs)
manage to call / receive call thru pstn (telekom malaysia)
i have problem with hangup signal.

one ip phone with extension => 101 register to asterisk
one telekom line (5512xxxx) in fxo port 1.
one mobilephone 01x-xxxxxxx (telco number) / other fixed line

  1. 101 call 01x-xxxxxxx and pickup then either one hangup first = no problem
  2. 01x-xxxxxxx call 5512xxxx wil route to 101 and pick up
    =>if 01x-xxxxxxx hang up first then 101 will get busy tone and hang up = no problem
    =>if 101 hangup first then 01x-xxxxxxx wil not hang up and get echo = anyone out there please help…

dahdi conf used

;;hanguponpolarityswitch=yes <-- tried this but no luck at all…
busydetect=yes
buycount=4
and also tried changing the
loopstart, kewlstart and groundstart signal … but stil no luck…

By hand phone, do you mean mobile phone (i.e. 手機)?

In the UK, and quite probably in Malaysia, mobile phone networks operate an either party clear policy, but landline networks operate a calling party clear policy, with a fallback to do called party clearing after a few minutes. The reasoning behind this is that the called party may have multiple extensions on one line and might want to hang up one and re-answer on a better placed one.

On SS7, the hangup and re-answer events are actualy signalled, although I don’t know if they are passed to non-PTT systems. On analogue systems, you will only see the final clear.

You may find that a line that is explicitly specified for PABX use will have called party clearing, but this will be something only the PTT can control.

yes is mobile phone… edited the first post…
ehh not quite understand from ur replied…
can u guide me what can i do and how do i solve this prob? :question:

If I understand what you are doing correctly, you need to talk to the telephone company providing the FXO line and ask them if it is possible to configure the line for either party clearing. It might not be, or you may have to rent a more expensive sort of line.

meaning that nothing can be done from d asterisk server? or i have to cal them up ask for line party clearing? how do i ask them? just state party clearing? they wil understand?? i dont think those cust serv will understand so much technical stuf…

You need to distinguish between called party clearing, which is normal for the PSTN and either party clearing.

If you have a domestic line, or possibly a business direct exchange line, it is probable that only calling party clearing is available, and the customer service people will be working from scripts and won’t understand you.

If you have a line intended for business PABX use, I would really hope that the customer service people would be able to work out what you are talking about, even if they can’t actually provide it, give or take any language issues.

Incidentally, if your PSTN has a calling party clearing policy, you should be able to demonstrate it by calling between two normal PSTN phones and temporarily hanging up the called one.

i just called them i dont think they understand my question…
from customer service transfer to technical support then pabx technical support…
all the answer i get is upgrade my line to pabx line (hunting line) cause my line is just a normal (home use) line.
do u think if i upgraded the line will fix my problem??

guess what?? i found the main problem. the test the problem by mobile calling the pstn line connected to analog phone i got the same problem… when analog phone hang up my mobile phone still connected to the line if i pickup d analog phone i stil able to talk to my mobile phone… so i guess the pstn line giving me problem…

I can’t say for certain that a hunt line will solve the problem, but I think it is likely.

Here is another reference to this phenomenom, in this case from Hong Kong:

ofta.gov.hk/en/ad-comm/tsac/ … 8p09a2.pdf

See item 8.6.

This goes into more detail:

ofta.gov.hk/en/standards/hkt … ta2017.pdf

but note that the ability to complete the clear:

  1. needs cooperation from the calling party;
  2. may require the calling party to have ISDN.

I don’t see any option for an analogue called party to force a Release in the Hong Kong document, but I only looked quickly and it may not be the most appropriate one.

thanks david ur info really help :smile:
i got another problem could help me on this?

a pstn line connected to fxo port 2. and both analog phone connected with extension 333 (port 3) and 444 (port 4)
when i call into d pstn line using different pstn line or mobile phone why am i receiving caller id from 444?
if i unregister both analog phone and why i get “unknown” why i dont see my mobile number as the caller id??

and also for outboundcall when i hit 9 follow by number i wan to dial let say. i dial 90123456789 i have to wait for like 5 seconds to ring that number is this normal?
thanks…

oh ya one more last question currently i got 2 fxo let say i got 2 pstn line how i manage d inbound cal?
i wan manage it by fxo port one call to ext 111 and fxo port 2 to ext 222… can we do that?