Asterisk Box -- two of the four trunk lines dont hang up

[mxc]

Hi all,

We have installed our asterisk server and have 3 incoming lines and 1 adsl. We originally had just two incoming lines while testing the system and all was fine. We added 2 more fxo ports to our digium card and now it seems that two of the incoming trunks dont hang up - at least for incoming calls. I notice on the flash operator panel that the lines just show a connected status.

Any one got any ideas on this?

[BradDC]

The problem exists in getting the FXO to detect that the caller has hung up.

Cisco's explanation

Understand the FXO Disconnect Problem

When loop-start signaling is used, a router's FXO interface looks like a phone to the switch (private branch exchange (PBX), public switched telephone network (PSTN), Key-System) it connects to. The FXO interface closes the loop to indicate off-hook. The switch always provides a battery so there is no disconnect supervision from the switch side. Since a switch expects a phone user (example of an FXO interface) to hang up the phone when the call is terminated (on either side), it also expects the FXO port on the router to hang-up. This "human intervention" is not built into the router. The FXO port expects the switch to tell it when to hang-up (or remove the battery to indicate on-hook). Because of this, there is no guarantee that a near-end or far-end FXO port disconnects the call once either end of the call hangs-up.

The most common symptoms of this problem are phones that continue to ring when the caller has cleared, or FXO ports that remain busy after the previous call should have been cleared.

Common Scenarios

As a simple rule-of-thumb, if the local router has an FXO port and it originates the call out of an FXO port, it has control over that call and can provide the local disconnect. If the local router has an FXO port and it receives the call, it requires that the connected switch provide this disconnect signal.

http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml#topic 1

Also take a look at this http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicem ail.html

[BradDC]

oh... and to double check... are you sure that they are all FXO modules (red)?
http://www.digitnetworks.com/store/images/fxo_module.jpg

[mxc]

yeah they are red. It seems to be the same trunks though that have the problem. Trunk 1 and 4 behave as they should but the trunks for 2 and 3 dont detect the hangup. Does this mean that for outgoing calls one could end up with a huge telkom bill if this happened on a Friday and you only found it on a Monday?

Scary stuff.

[ant101]

mxc, with a bit of tweaking you can survive!
For software I would check the following:
Telkom uses two switch types in their network, Alcatel switches - mainly in the Western Cape, and Siemens switches further north. If you are situated in Cape Town, Alcatel, is likely to work but we found that Siemens setup works better.
Make sure of the following is listed in your indications.conf file.
Uncomment the ones you prefer with a “;”. Just below the [general] heading change the country=us to country=za. Doing this will tell Asterisk to use the [za] context instead of the USA one.

[general]
country=za
[za]
description = South Africa
ringcadance = 400,200,400,2000
; dial/ring/callwaiting for the Siemens switches:
dial = 400*33
ring = 400*33/400,0/200,400*33/400,0/2000
callwaiting =
400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/….
; dial/ring/callwaiting for the Alcatel switches:
;dial = 383+417
;ring = 383+417/400,0/200,383+417/400,0/2000
;callwaiting =
383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383….
congestion = 400/250,0/250
busy = 400/500,0/500
dialrecall = 350+440

Since you did not have the problem when you had the 2 lines, I would also check your hardware. If you are running FXS ports, then make sure they have one of the white molex power plugs to them. I have heard of some smaller power supplys in pc's not being able to cope with many fxs ports. (upgrade of pc supply is needed to solve this.) Dodgey supply=dodgey operation.
I also remember reading about the silence threshhold which can be set, but I think this was only for the asterisk vociemail -so it could detect when a caller hangs up, and not the actual card ports.

[mxc]

thanks --- I am using trixbox which seems to not have the sa settings. Also I dont think there will be a billing problem if the line remains locked. It seems that it always detects sip hand ups. ie. outgoing calls. Its only on incoming calls that the line does not drop if the caller hangsup before connecting to a sip phone.

[Gerry]

I have to fully agree with all the posts on this topic.

1. Make sure your atx power supply powers the Fxs modules. Fxo usually has no problem as it does not need to generate tones and voltages.
2. Ensure you have the latest fxotune executable - just this will perhaps take care of 90% percent of your problems.
3. Re compile your zaptel drivers to the latest digium or svn it (current 1.2.7)

[Nickste]

Hi all,

Found this post while searching for a solution to the same problem.
Checkout this page: http://www.jumpingbean.co.za/blogs/mark/asterisk_south_africa and note that if you are using asterisk 1.6, you must edit the chan_dahdi.conf file instead of the zapata.conf file.

Cheers,
Nick

[Madman88]

Hi all,

Found this post while searching for a solution to the same problem.
Checkout this page: http://www.jumpingbean.co.za/blogs/mark/asterisk_south_africa and note that if you are using asterisk 1.6, you must edit the chan_dahdi.conf file instead of the zapata.conf file.

Cheers,
Nick

:D Thanks for that!!

[Nickste]

No problem :)

[Madman88]

Ok, so it worked for a bit, now its keeping the lines open again...

[Nickste]

Hmmmm.. that's strange. Maybe try setting back to old settings, applying, and then readding the SA settings?

[Madman88]

ok, I kinda did that and its working now...

I changed the settings to us, rebooted it, changed them back to za and rebooted agan.

Happy Again!! :D