Discussion:
Calls take 30+ seconds to clear down after hanging up
(too old to reply)
Mark Carver
2023-11-04 15:38:33 UTC
Permalink
As title. Made a couple of test calls to my mobile, and POTs landline,
the moment the remote end phone starts ringing I hang up, by the time
the connection is actually dropped, the mobile etc has clicked into
Voicemail, and 14p has gone.

Phone is Grandstream 1610, VoIP provider is Sipgate.

Where to a start looking to solve ?

Be gentle, I'm new to all this !
Woody
2023-11-04 16:16:23 UTC
Permalink
Post by Mark Carver
As title. Made a couple of test calls to my mobile, and POTs landline,
the moment the remote end phone starts ringing I hang up, by the time
the connection is actually dropped, the mobile etc has clicked into
Voicemail, and 14p has gone.
Phone is Grandstream 1610, VoIP provider is Sipgate.
Where to a start looking to solve ?
Be gentle, I'm new to all this !
What RTP and UTP ports are you using? Do you have Stun active?
Woody
2023-11-04 16:15:54 UTC
Permalink
Post by Mark Carver
As title. Made a couple of test calls to my mobile, and POTs landline,
the moment the remote end phone starts ringing I hang up, by the time
the connection is actually dropped, the mobile etc has clicked into
Voicemail, and 14p has gone.
Phone is Grandstream 1610, VoIP provider is Sipgate.
Where to a start looking to solve ?
Be gentle, I'm new to all this !
What RTP and UTP ports are you using? Do you have Stun active?
Woody
2023-11-04 16:17:51 UTC
Permalink
Post by Mark Carver
As title. Made a couple of test calls to my mobile, and POTs landline,
the moment the remote end phone starts ringing I hang up, by the time
the connection is actually dropped, the mobile etc has clicked into
Voicemail, and 14p has gone.
Phone is Grandstream 1610, VoIP provider is Sipgate.
Where to a start looking to solve ?
Be gentle, I'm new to all this !
Have you looked through the setup pages on the Sipgate web site?
Are you using the Stun server?
What values are you using for RTP and UDP?
Mark Carver
2023-11-04 16:26:56 UTC
Permalink
Post by Woody
Post by Mark Carver
As title. Made a couple of test calls to my mobile, and POTs landline,
the moment the remote end phone starts ringing I hang up, by the time
the connection is actually dropped, the mobile etc has clicked into
Voicemail, and 14p has gone.
Phone is Grandstream 1610, VoIP provider is Sipgate.
Where to a start looking to solve ?
Be gentle, I'm new to all this !
Have you looked through the setup pages on the Sipgate web site?
Yes, I've entered their settings for the model of phone
Post by Woody
Are you using the Stun server?
Yep stun.sipgate.net
Post by Woody
What values are you using for RTP and UDP?
Local RTP Port 49104
Local RTP Port Range 200

UDP Dunno, there's no where in the phone config to enter a value etc ?
Woody
2023-11-04 18:54:14 UTC
Permalink
Post by Mark Carver
Post by Woody
Post by Mark Carver
As title. Made a couple of test calls to my mobile, and POTs
landline, the moment the remote end phone starts ringing I hang up,
by the time the connection is actually dropped, the mobile etc has
clicked into Voicemail, and 14p has gone.
Phone is Grandstream 1610, VoIP provider is Sipgate.
Where to a start looking to solve ?
Be gentle, I'm new to all this !
Have you looked through the setup pages on the Sipgate web site?
Yes, I've entered their settings for the model of phone
Post by Woody
Are you using the Stun server?
Yep  stun.sipgate.net
Post by Woody
What values are you using for RTP and UDP?
Local RTP Port  49104
Local RTP Port Range  200
UDP Dunno, there's no where in the phone config to enter a value etc ?
That's what happens when you do it from memory!

It should have been the SIP section SIP port which should be 43160

Stun should be stun.sipgate.net:10000

Have you set the impedance match which should be 310nF||370+620

If you let me have your email address I'll send you the config of my
PAP2T which works a treat on Sipgate. It will be different from your
1610 but it may give you some ideas.
I also have an app on my (Android) phone for Sipgate the config of which
I can send and you can use to see if it performs any differently to your
1610.
Mark Carver
2023-11-05 09:16:18 UTC
Permalink
Post by Woody
It should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by Woody
Stun should be stun.sipgate.net:10000
So Port 10000 ?
Post by Woody
Have 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
Woody
2023-11-05 11:45:39 UTC
Permalink
Post by Mark Carver
Post by Woody
It should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by Woody
Stun should be stun.sipgate.net:10000
So Port 10000 ?
Post by Woody
Have 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.

Final silly question: is your codec set as G729a? Don't know if it will
make any difference but..... Also check all the audio settings.
Mark Carver
2023-11-05 12:40:06 UTC
Permalink
Post by Woody
Post by Mark Carver
Post by Woody
It should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by Woody
Stun should be stun.sipgate.net:10000
So Port 10000 ?
Post by Woody
Have 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 Woody
Final 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
Woody
2023-11-05 12:42:46 UTC
Permalink
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
It should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by Woody
Stun should be stun.sipgate.net:10000
So Port 10000 ?
Post by Woody
Have 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 Woody
Final 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.....
Mark Carver
2023-11-05 13:44:18 UTC
Permalink
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
It should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by Woody
Stun should be stun.sipgate.net:10000
So Port 10000 ?
Post by Woody
Have 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 Woody
Final 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.
Woody
2023-11-05 15:36:57 UTC
Permalink
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
It should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by Woody
Stun should be stun.sipgate.net:10000
So Port 10000 ?
Post by Woody
Have 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 Woody
Final 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.
Mark Carver
2023-11-05 16:26:28 UTC
Permalink
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
It should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by Woody
Stun should be stun.sipgate.net:10000
So Port 10000 ?
Post by Woody
Have 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 Woody
Final 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.
Woody
2023-11-05 19:44:07 UTC
Permalink
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
It should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by Woody
Stun should be stun.sipgate.net:10000
So Port 10000 ?
Post by Woody
Have 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 Woody
Final 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.

I'm looking to escape VM at the end of the year and go FTTP (Cityfibre
outside my gate) and will probably use one of these boxes.
Mark Carver
2023-11-06 08:53:49 UTC
Permalink
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
It should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by Woody
Stun should be stun.sipgate.net:10000
So Port 10000 ?
Post by Woody
Have 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 Woody
Final 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.
Woody
2023-11-06 08:59:40 UTC
Permalink
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
Post by Mark Carver
Post by Woody
It should have been the SIP section SIP port which should be 43160
That's currently set to 49160
Post by Woody
Stun should be stun.sipgate.net:10000
So Port 10000 ?
Post by Woody
Have 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 Woody
Final 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.
Yes, I felt the same about the Basic service withdrawal and waited for
them to let me know - and waited, and waited, and I'm still waiting.
Mind you I have had an account with them for at least 15 years or more
so maybe I have escaped the chop even though I rarely use it!

I also have an account with what was voip.co.uk, now known as First
Europe. They still run a basic service and their call rates are also
comparable or sometimes cheaper than Sipgate.

By the way, you can switch off the shouty message about call cost every
time you make a call. Just log in and find it in the settings.

Good luck.
Bob Eager
2023-11-06 10:36:58 UTC
Permalink
Post by Mark Carver
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)
I just ported the number to my current ITSP (AAISP). Sipgate refunded the
balance of my credit very quickly. Job done.
Woody
2023-11-06 10:52:08 UTC
Permalink
Post by Bob Eager
Post by Mark Carver
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)
I just ported the number to my current ITSP (AAISP). Sipgate refunded the
balance of my credit very quickly. Job done.
In that respect Sipgate are very good, as they are if you discover a
service fault albeit you usually have to call their business service to
get a real person.

In an earlier post I mentioned voip.co.uk now called First Europe. They
are equally helpful and prompt.

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