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Problem with VOIP adaptor + Localphone.com + Internet Via Satellite
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Anti-Spam
Mar 14, 2013, 4:29:07 PM3/14/13
to
I am having problems setting up a Handytone HT502 with Localphone.com.
The internet connection is coming in, via Internet Via Satellite, with
a small amount of Latency, if thats relevant?
Also, if its relevant? the grandstream is set to NAT router mode for
the Access Point (Edimax) which follows it.
I have set it up, according to the nearest grandstream model set-up
guide, localphone have on their web site.
I am getting three problems listed in order of importance.
1. On any incoming call on the VOIP system, the POT handset rings
about 2 times. If you manage to pick the phone up, before the end of
the two rings, then you can use the phone as normal with no problems
(we are aware of). If you do not pick up within the two rings, the
phone stops ringing and the caller stops hearing a ringing tone, and
gets an unobtainable tone.
2. When making outgoing calls on the VOIP system, the phone works as
it should only if the person called, picks the phone up. But if the
person called, does not pick up, and the caller puts the handset down,
it does not stop the phone ringing at the person called end (we have
only tried this with a mobile). If you pick up the VOIP phone, it now
gives the dial tone (even though the called persons phone is still
ring) eventually the called persons phone, times out and stops
ringing.
3. Here in Spain, there are phone systems given to people (who live in
rural areas where there are no phone lines) by the national phone
company (telefonica) which act like normal phones, but are actually
connected, via the telefonica mobile phone network. These TRAC phones,
as they are called, are charged at normal landline phone call rates.
We have found (by accident) that if the VOIP phone is rung by a TRAC
system phone, the message comes back to the caller that the number is
unobtainable, we tried it on two different TRAC phones and got the
same result.
The internet connection is coming in, via Internet Via Satellite, with
a small amount of Latency, if thats relevant?
Also, if its relevant? the grandstream is set to NAT router mode for
the Access Point (Edimax) which follows it.
I have set it up, according to the nearest grandstream model set-up
guide, localphone have on their web site.
I am getting three problems listed in order of importance.
1. On any incoming call on the VOIP system, the POT handset rings
about 2 times. If you manage to pick the phone up, before the end of
the two rings, then you can use the phone as normal with no problems
(we are aware of). If you do not pick up within the two rings, the
phone stops ringing and the caller stops hearing a ringing tone, and
gets an unobtainable tone.
2. When making outgoing calls on the VOIP system, the phone works as
it should only if the person called, picks the phone up. But if the
person called, does not pick up, and the caller puts the handset down,
it does not stop the phone ringing at the person called end (we have
only tried this with a mobile). If you pick up the VOIP phone, it now
gives the dial tone (even though the called persons phone is still
ring) eventually the called persons phone, times out and stops
ringing.
3. Here in Spain, there are phone systems given to people (who live in
rural areas where there are no phone lines) by the national phone
company (telefonica) which act like normal phones, but are actually
connected, via the telefonica mobile phone network. These TRAC phones,
as they are called, are charged at normal landline phone call rates.
We have found (by accident) that if the VOIP phone is rung by a TRAC
system phone, the message comes back to the caller that the number is
unobtainable, we tried it on two different TRAC phones and got the
same result.
Anti-Spam
Mar 15, 2013, 3:08:50 AM3/15/13
to
(running latest firmware), if it helps. They can be found at
www.satseekers.net/netseekers/voipproblem/corel075.jpg
all the way to
www.satseekers.net/netseekers/voipproblem/corel088.jpg
David Woolley
Mar 15, 2013, 4:33:16 AM3/15/13
to
Anti-Spam wrote:
> I am having problems setting up a Handytone HT502 with Localphone.com.
> The internet connection is coming in, via Internet Via Satellite, with
> a small amount of Latency, if thats relevant?
If this uses geosynchronous satellites, you won't get less than medium
> I am having problems setting up a Handytone HT502 with Localphone.com.
> The internet connection is coming in, via Internet Via Satellite, with
> a small amount of Latency, if thats relevant?
latency.
>
> Also, if its relevant? the grandstream is set to NAT router mode for
> the Access Point (Edimax) which follows it.
The risk here is that it also munging the SIP headers.
> Also, if its relevant? the grandstream is set to NAT router mode for
> the Access Point (Edimax) which follows it.
>
>
> 1. On any incoming call on the VOIP system, the POT handset rings
> about 2 times. If you manage to pick the phone up, before the end of
> the two rings, then you can use the phone as normal with no problems
> (we are aware of). If you do not pick up within the two rings, the
> phone stops ringing and the caller stops hearing a ringing tone, and
> gets an unobtainable tone.
This sounds as though interim responses are not getting through. That
>
> 1. On any incoming call on the VOIP system, the POT handset rings
> about 2 times. If you manage to pick the phone up, before the end of
> the two rings, then you can use the phone as normal with no problems
> (we are aware of). If you do not pick up within the two rings, the
> phone stops ringing and the caller stops hearing a ringing tone, and
> gets an unobtainable tone.
is weird. Also, I don't think a normal SIP system should abort the call
after that short a time without a response. Does the caller get
ringback tone when the adapter is powered down. That would indicate
that the ringback is being generated before the network has had
confirmation of ringing. If you get immediate unobtainable, this test
may not be valid, because a non-SIP mechanism is actively reporting the
loss of connection.
>
> 2. When making outgoing calls on the VOIP system, the phone works as
> it should only if the person called, picks the phone up. But if the
> person called, does not pick up, and the caller puts the handset down,
> it does not stop the phone ringing at the person called end (we have
adaptor or the central system failing to include tags, or insisting on
them when the protocol doesn't actually require them.
Ideally you need a packet capture and a trace of the SIP protocol, to
see what is happening.
> only tried this with a mobile). If you pick up the VOIP phone, it now
> gives the dial tone (even though the called persons phone is still
> ring) eventually the called persons phone, times out and stops
> ringing.
network. The adaptor can safely return it immediately, as it can start a
new SIP call whilst still retrying the CANCEL on the old one.
>
> These TRAC phones,
> as they are called, are charged at normal landline phone call rates.
> We have found (by accident) that if the VOIP phone is rung by a TRAC
> system phone, the message comes back to the caller that the number is
> unobtainable, we tried it on two different TRAC phones and got the
> same result.
This sounds like it is due to commercial, rather than purely technical,
> These TRAC phones,
> as they are called, are charged at normal landline phone call rates.
> We have found (by accident) that if the VOIP phone is rung by a TRAC
> system phone, the message comes back to the caller that the number is
> unobtainable, we tried it on two different TRAC phones and got the
> same result.
issues.
M & S
Mar 15, 2013, 7:07:35 AM3/15/13
to
>Anti-Spam wrote:
>> I am having problems setting up a Handytone HT502 with Localphone.com.
>> The internet connection is coming in, via Internet Via Satellite, with
>> a small amount of Latency, if thats relevant?
>
>If this uses geosynchronous satellites, you won't get less than medium
>latency.
Its using the new service from Tooway, which now uses Eutelsat9
>> I am having problems setting up a Handytone HT502 with Localphone.com.
>> The internet connection is coming in, via Internet Via Satellite, with
>> a small amount of Latency, if thats relevant?
>
>If this uses geosynchronous satellites, you won't get less than medium
>latency.
>>
>> Also, if its relevant? the grandstream is set to NAT router mode for
>> the Access Point (Edimax) which follows it.
>
>The risk here is that it also munging the SIP headers.
Grandstream HT502.
Grandstream HT502 set to NAT issuing DHCP IP addresses (in a different
subnet) to a Edimax Access point. I presumed that the VOIP traffic
would be processed before the NAT router section, in which case do we
need to do something with NAT traversal? or do you not mean this?
>>
>>
>> 1. On any incoming call on the VOIP system, the POT handset rings
>> about 2 times. If you manage to pick the phone up, before the end of
>> the two rings, then you can use the phone as normal with no problems
>> (we are aware of). If you do not pick up within the two rings, the
>> phone stops ringing and the caller stops hearing a ringing tone, and
>> gets an unobtainable tone.
>
>This sounds as though interim responses are not getting through. That
>is weird. Also, I don't think a normal SIP system should abort the call
>after that short a time without a response. Does the caller get
>ringback tone when the adapter is powered down. That would indicate
>that the ringback is being generated before the network has had
>confirmation of ringing. If you get immediate unobtainable, this test
>may not be valid, because a non-SIP mechanism is actively reporting the
>loss of connection.
straight away. I have been in touch with Localphone, and here is what
they said
"I believe the latency issues are what is causing your problems. If we
do not get the correct SIP code response in a timely manner you will
experience the problems you are facing.
You could try reducing the amount of seconds between Registration
Retry's, this is usual set at around 3600 try setting it to 360.
Also you may want to set your VoIP device on your routers DMZ to help
prevent any NAT issues."
David, does that sound about right?
At the moment I have (and I think this is the one he means) "SIP
registration Failure Retry wait time" set to 20 seconds.
>
>>
>> 2. When making outgoing calls on the VOIP system, the phone works as
>> it should only if the person called, picks the phone up. But if the
>> person called, does not pick up, and the caller puts the handset down,
>> it does not stop the phone ringing at the person called end (we have
>
>That indicates a problem with CANCEL. There can be issues with the
>adaptor or the central system failing to include tags, or insisting on
>them when the protocol doesn't actually require them.
>
>Ideally you need a packet capture and a trace of the SIP protocol, to
>see what is happening.
Is there some freeish software to do that?
>>
>> 2. When making outgoing calls on the VOIP system, the phone works as
>> it should only if the person called, picks the phone up. But if the
>> person called, does not pick up, and the caller puts the handset down,
>> it does not stop the phone ringing at the person called end (we have
>
>That indicates a problem with CANCEL. There can be issues with the
>adaptor or the central system failing to include tags, or insisting on
>them when the protocol doesn't actually require them.
>
>Ideally you need a packet capture and a trace of the SIP protocol, to
>see what is happening.
>
>> only tried this with a mobile). If you pick up the VOIP phone, it now
>> gives the dial tone (even though the called persons phone is still
>> ring) eventually the called persons phone, times out and stops
>> ringing.
>
>Dialtone will be generated by the adaptor without talking to the
>network. The adaptor can safely return it immediately, as it can start a
>new SIP call whilst still retrying the CANCEL on the old one.
>>
>> These TRAC phones,
>> as they are called, are charged at normal landline phone call rates.
>> We have found (by accident) that if the VOIP phone is rung by a TRAC
>> system phone, the message comes back to the caller that the number is
>> unobtainable, we tried it on two different TRAC phones and got the
>> same result.
>
>This sounds like it is due to commercial, rather than purely technical,
>issues.
satellite internet equipment. This is a TRAC phone calling a normal
Spanish national phone number are you suggesting that it is detecting
that its a VOIP system being called or do you mean that the number is
flagged as being from a VOIP service provider?
Thanks for your time and energy on helping me, appreciated.
Mark Scotford
Castellon
M & S
Mar 15, 2013, 7:09:40 AM3/15/13
to
Sorry about the identity crisis, I updated the software of my Agent
software, now it keeps asking me for an email address to use to send
usenet messages? will change software soon, sorry.
software, now it keeps asking me for an email address to use to send
usenet messages? will change software soon, sorry.
Anti-Spam
Mar 15, 2013, 7:17:01 AM3/15/13
to
but the "'Registration Expiration' try reducing this to 15 minutes."
this is currently set to 60 minutes
Does that sound about right?
Dave Saville
Mar 15, 2013, 9:29:51 AM3/15/13
to
On Fri, 15 Mar 2013 11:07:35 UTC, M & S <m...@m.com> wrote:
> Grandstream HT502 set to NAT issuing DHCP IP addresses (in a different
> subnet) to a Edimax Access point. I presumed that the VOIP traffic
> would be processed before the NAT router section, in which case do we
> need to do something with NAT traversal? or do you not mean this?
Yes you do. If any SIP device is behind NAT then you need either:
> Grandstream HT502 set to NAT issuing DHCP IP addresses (in a different
> subnet) to a Edimax Access point. I presumed that the VOIP traffic
> would be processed before the NAT router section, in which case do we
> need to do something with NAT traversal? or do you not mean this?
A SIP ALG in the router - Usually buggy.
Use a STUN server
Use a Proxy server
The problem is that the SIP protocol puts the device address *inside*
the packets so "normal" NAT does not see/change them. You get outgoing
but no incomming as the other end does not know where to send the
response.
HTH
--
Regards
Dave Saville
Andrew Benham
Mar 15, 2013, 1:24:29 PM3/15/13
to
configured to tell them what the external IP address is. Obviously only
of use if you have a static external IP address but it saves messing
about with (as you say) often broken SIP ALGs, or STUN servers.
alexd
Mar 15, 2013, 4:02:49 PM3/15/13
to
Dave Saville (for it is he) wrote:
> Yes you do. If any SIP device is behind NAT then you need either:
> A SIP ALG in the router - Usually buggy.
> Use a STUN server
Surely ITSPs have got workarounds for this by now - all they need to do is
> Yes you do. If any SIP device is behind NAT then you need either:
> A SIP ALG in the router - Usually buggy.
> Use a STUN server
reply to the source IP address rather than what's in the SIP headers?
> Use a Proxy server
I think most people fall into this category. Set their SIP device to use
their ITSPs server as their SIP proxy.
--
<http://ale.cx/> (AIM:troffasky) (UnSoEs...@ale.cx)
20:00:14 up 11:56, 2 users, load average: 0.70, 0.59, 0.65
Qua illic est reprehendit, illic est a vindicatum
Graham.
Mar 15, 2013, 4:10:24 PM3/15/13
to
On Fri, 15 Mar 2013 20:02:49 +0000, alexd <trof...@hotmail.com>
wrote:
>Dave Saville (for it is he) wrote:
>
>> Yes you do. If any SIP device is behind NAT then you need either:
>> A SIP ALG in the router - Usually buggy.
>> Use a STUN server
>
>Surely ITSPs have got workarounds for this by now - all they need to do is
>reply to the source IP address rather than what's in the SIP headers?
>
>> Use a Proxy server
>
>I think most people fall into this category. Set their SIP device to use
>their ITSPs server as their SIP proxy.
Can you explain the alternatives and their advantages?
--
Graham.
%Profound_observation%
wrote:
>Dave Saville (for it is he) wrote:
>
>> Yes you do. If any SIP device is behind NAT then you need either:
>> A SIP ALG in the router - Usually buggy.
>> Use a STUN server
>
>Surely ITSPs have got workarounds for this by now - all they need to do is
>reply to the source IP address rather than what's in the SIP headers?
>
>> Use a Proxy server
>
>I think most people fall into this category. Set their SIP device to use
>their ITSPs server as their SIP proxy.
--
Graham.
%Profound_observation%
alexd
Mar 15, 2013, 4:29:48 PM3/15/13
to
M & S (for it is he) wrote:
> "I believe the latency issues are what is causing your problems.
Yes. That sounds likely, and I've seen similar symptoms with proprietary
> "I believe the latency issues are what is causing your problems.
VoIP over high-latency links.
> You could try reducing the amount of seconds between Registration
> Retry's, this is usual set at around 3600 try setting it to 360.
instead of every hour won't stop you from answering a call after n rings. If
your problem was that incoming calls stopping working after ~6 minutes
between calls, then this might be a useful fix.
> Is there some freeish software to do that?
some way to physically direct the traffic you want to capture to your
machine running Wireshark in order to do the capture, if the kit you're
using can't do packet captures itself. An ethernet hub will do that, but
they're pretty thin on the ground nowadays, decent managed switches will do
port mirroring, but again I doubt you've got one.
http://wiki.wireshark.org/CaptureSetup/Ethernet#Switched_Ethernet
> This is a TRAC phone calling a normal
> Spanish national phone number are you suggesting that it is detecting
> that its a VOIP system being called or do you mean that the number is
> flagged as being from a VOIP service provider?
in Spain to know if Telefonica are allowed to get away with this, but I
doubt the BT would be allowed to block calls to Gamma or Magrathea numbers.
20:03:55 up 11:59, 2 users, load average: 0.53, 0.58, 0.63
alexd
Mar 15, 2013, 6:21:29 PM3/15/13
to
internet - make calls directly to SIP the endpoint you want to call.
In an ideal world, we [our software] would look someone up in a central
directory [DNS] and connect straight to them to make a call, but I think
that's probably likely to lead to a world of spammy pain.
Most ITSP users will be using their ITSPs proxy to make calls to the PSTN.
22:12:28 up 14:08, 2 users, load average: 0.56, 0.53, 0.57
Anti-Spam
Mar 17, 2013, 4:22:24 PM3/17/13
to
sake, just so we are clear. My system is Satellite Internet Modem,
feeds Grandstream HT502 (Incorporating NAT), (WAN side on DHCP talking
to Satellite internet modem) (LAN side on different Subnet, talking to
a Belkin Internet Access Point) I assumed that the VOIP section of the
VOIP adapter, would be on the WAN side of things, so not affected by
the NAT, or am I being a Twat?
Dave Saville
Mar 18, 2013, 9:01:25 AM3/18/13
to
So where is the VOIP adaptor?
--
Regards
Dave Saville
Message has been deleted
Dave Saville
Mar 18, 2013, 12:02:40 PM3/18/13
to
On Mon, 18 Mar 2013 13:23:38 UTC, "Anthony R. Gold"
<not-fo...@ahjg.co.uk> wrote:
> On Mon, 18 Mar 2013 13:01:25 +0000 (UTC), "Dave Saville"
> http://www.grandstream.com/index.php/products/ip-voice-telephony/consumer-analog-telephone-adaptors/ht502
Page 25 of the 502 manual gives the impression that VOIP is behind NAT
--
Regards
Dave Saville
<not-fo...@ahjg.co.uk> wrote:
> On Mon, 18 Mar 2013 13:01:25 +0000 (UTC), "Dave Saville"
Page 25 of the 502 manual gives the impression that VOIP is behind NAT
--
Regards
Dave Saville
Anti-Spam
Mar 21, 2013, 3:56:31 AM3/21/13
to
On Mon, 18 Mar 2013 16:02:40 +0000 (UTC), "Dave Saville"
Thanks lads, I will drive up to the customers place and turn off the
NAT in the HT502, and put it into bridge mode, instead.
I cant replicate the problem here in the office, because we dont have
a Satellite Internet connection.
NAT in the HT502, and put it into bridge mode, instead.
I cant replicate the problem here in the office, because we dont have
a Satellite Internet connection.
Anti-Spam
Mar 24, 2013, 8:50:18 PM3/24/13
to
Sipgate.co.uk settings on same kit and it worked fine (apart from the
expected slight delay in comms). Will get onto Localphone.com today.
Anti-Spam
Apr 9, 2013, 3:10:18 AM4/9/13
to
On Fri, 15 Mar 2013 20:29:48 +0000, alexd <trof...@hotmail.com>
wrote:
I put the customer onto www.voiptalk.co.uk (with no special settings)
and they are now very happy with their new VOIP phone, albeit with a
slight delay (not half as bad as I was expecting) in communications.
They got a refund from localphone.com, with a -10% admin fee.
I am still with www.localphone.com (along with 3 other providers) for
a Romanian incoming phone number, as they were the cheapest for
International Numbers. I am Not on Internet via satellite, I am using
a Fritzbox 7170 on a ADSL line. All my other services from the other 3
providers work very well, except the localphone.com one. I do not use
them for outgoing calls, but on incoming calls, the phone rings a few
times and then stops (if I do not pick up), the caller then gets a
unobtainable tone. Technical support can only tell me its a router
problem?
wrote:
and they are now very happy with their new VOIP phone, albeit with a
slight delay (not half as bad as I was expecting) in communications.
They got a refund from localphone.com, with a -10% admin fee.
I am still with www.localphone.com (along with 3 other providers) for
a Romanian incoming phone number, as they were the cheapest for
International Numbers. I am Not on Internet via satellite, I am using
a Fritzbox 7170 on a ADSL line. All my other services from the other 3
providers work very well, except the localphone.com one. I do not use
them for outgoing calls, but on incoming calls, the phone rings a few
times and then stops (if I do not pick up), the caller then gets a
unobtainable tone. Technical support can only tell me its a router
problem?
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