Hi,
I have a TDM410P with one fxo and I have configured a day/night message with hungup at the end.
In the asterisk CLI all works and asterisk try to terminate the call with the HANGUP command to the fxo but the call don’t terminate and remain active.
How can I solve this problem ?
I have installed the last DAHDI drivers and update asterisk.
Try with these options in chan_dahdi.conf:
hanguponpolarityswitch
busydetect
Ask your PSTN provider. What you want may not be possible.
In the UK, on analogue landlines, the called party cannot clear the call (it will clear after a few minutes). This is done so that the called party can put down the extension on which they answered and re-answer on a more appropriate extension.
I have a pure VOIP platform without any fxo/fxs. Whenever the SIP phone is hung up, the connection remains active. I tried commenting out all the lines in musiconhold.conf because before the call used to stay on hold. Now, whenever the SIP phone is hung up it says “Music class default requested but no musiconhold loaded.” and the line remains active for long time. Sometime the call remains active for few minutes even though the SIP phone is hangged up. Also, when I press “FLASH” button to make another call, the call old line goes into hold status (off course without music because everthing on musiconhold.conf is disable).
NOW:
How do I make the call end as soon as I hang the phone?
How do make sure the call doesn’t hold when I press “FLASH” button to redial the same number or dial another number?
My setting information:
Asterisk 1.6.2.7 on dedicated server
SIP Phone: Linksys PAP-2NA
[1111]
type=friend
username=1111
accountcode=1111
regexten=1111
callerid=1234567890
amaflags=billing
secret=secret
nat=yes
dtmfmode=RFC2833
qualify=yes
canreinvite=yes
disallow=all
allow=g729
allow=g723
allow=gsm
allow=ulaw
allow=alaw
host=dynamic
context=a2billing
rtptimeout=10
rtpholdtimeout=10
regseconds=0
cancallforward=yes
It would have been better to start a new thread.
On a SIP phone, the flash button handling is done by the phone, so that one is outside the scope of Asterisk.
For your hangup problem, you need to provide a SIP protocol trace, so we can understand what is going wrong.