Post by WoodyPost by Mark CarverPost by WoodyPost by Mark CarverPost by WoodyPost by Mark CarverPost by WoodyPost by Mark CarverPost by WoodyIt should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by WoodyStun should be stun.sipgate.net:10000
So Port 10000 ?
Post by WoodyHave you set the impedance match which should be 310nF||370+620
Impedance matching what to what exactly ? I'm even more
confused. It's just a totally self contained VoIP desk phone,
with nothing more than an Ethernet connection to my router ?
I should add that incoming calls work fine, as do calls to
Sipgate's own test numbers 10000, 10005, 10020 etc
https://teamhelp.sipgate.co.uk/hc/en-gb/articles/6275616468765-Grandstream-GXP-1100-1400-1600-and-2100-Series
Confirm port 10000
Sorry I missed that it is a standalone phone. The impedance match
is if you are using a VoIP ATA for a conventional phone to be
plugged into the FXS port(s).
I don't know if 49160 is any sort of digital mismatch but mine is
43160 which is in the same 'region' as the RTP range which I
think you said on yours is 43104-43120.
Tried youR settings. Made no difference at all. In fact I dialled
the number, waited for Mr Sipgate to say 'Only 1p or whatever a
min' and hung up before the remote phone even started ringing. It
then rang for 30 seconds.
Post by WoodyFinal silly question: is your codec set as G729a? Don't know if
it will make any difference but..... Also check all the audio
settings.
G.722
Try G729a - which on a PAP2T is default - you never know.....
Why would changing the codec have any effect on the cleardown delay ?
BTW when clearing down on an incoming VoIP call, the cleardown is
almost instant.
Just remembered: check your router settings and if it is not, disable
SIP ALG. This parameter causes more problems with SIP calls than
almost anything else.
Non disablble on my router unfortunately.
Oh well, VoIP's clearly not for me.
Doesn't matter, I'm only intending to use for incoming calls anyway
(and they work just fine)
I'm beginning to understand why the major ISPs implement as a
proprietary system inside their routers.
There's no way the great unwashed can navigate all this mucking around
with ports etc.
I have to agree Mark, but don't be put off. Get yourself a Cisco PAP2T
(be careful you get one with the Cisco (or Linksys) badge on it -
there's a lot of Chinese copying around!)
They are simple enough to set up, very reliable, and they work,
period.The 2T has two FXS (phone) sockets and you can dual config the
box so that you can use two phones on two accounts on the one box. I've
got two here at the side of me although only one is in use. I think they
cost me about £20 apiece.
You can also get a Gigaset GO-Box 100 which is a DECT base station and
combined VoIP and PSTN interface. Most £20-£40.
It's really not worth the expense and effort for me.
I've simply set up the Sipgate account to continue to have a 'landline'
number for incoming calls only after the point I go FTTP (or Plusnet
kill the PSTN service, which ever comes first, but it's going to be no
later than 24 months from now)
It's only for the benefit of about about half a dozen family nonagenarians.
Unless a miracle happens, I doubt I will be using the account beyond the
end of this decade.
As said incoming calls are fine. I've been signed up since June 2022,
but held back sticking on any credit until now, because Sipgate
announced a year ago they were ceasing the PAYG service. I've gave up
waiting this weekend, and lumped in a Tenner's worth of credit. It's
this that has highlighted the 'sticky' cleardown issue for outgoing
calls.