The configuration looks correct to the untrained eye, it says it is up
and running, shows my correct a/c number as the user name, its set to
sip.voipfone.net as suggested by their support with the right port
number, and it correctly tells me the number of messages waiting.
It makes outgoing calls fine.
However the phone connected to it doesn't ring when a call comes in,
I've tried a number of different phones, both cordless and corded but it
makes no difference.
>
>> However the phone connected to it doesn't ring when a call comes in,
> I've tried a number of different phones, both cordless and corded but it
> makes no difference.
>
> Any suggestions
On 12/05/2011 19:59, Blu wrote:
>
> I have a Patton micro ATA.
>
> The configuration looks correct to the untrained eye, it says it is up
> and running, shows my correct a/c number as the user name, its set to
> sip.voipfone.net as suggested by their support with the right port
> number, and it correctly tells me the number of messages waiting.
>
> It makes outgoing calls fine.
>
> However the phone connected to it doesn't ring when a call comes in,
> I've tried a number of different phones, both cordless and corded but it
> makes no difference.
>
> Any suggestions of things to try please
>
> Blu.
>
> Router is Linksys WRT54G
What happens if you pick up[1] the phone when it should be ringing, but
isn't - can you then hold a conversation with the calling party? Or do
you just get dial-tone, indicating that the call hasn't been connected?
[1] You could test this by ringing from a landline (or mobile) in the
next room (say).
--
Cheers,
Roger
____________
Please reply to Newsgroup. Whilst email address is valid, it is seldom
checked.
Blu <blu@green.com> wrote:
> I have a Patton micro ATA.
> The configuration looks correct [...]
> It makes outgoing calls fine.
> However the phone connected to it doesn't ring when a call comes in [...]
Try picking up the phone when you know it's (supposed to be) ringing.
If it answers then you need to look at how the ATA tells the phone to
ring. (Another poster has suggested a ringing device.)
On the other hand, if you just get a dial tone and the call isn't answered
then it's more likely to be a NAT issue, and you'll need to check your
Proxy/Registration timeout vs your firewall NAT timeout.
On 2011-05-13, I'm Old Gregg <mantrid.a@ukonline.invalid> wrote:
>
>>
>>> However the phone connected to it doesn't ring when a call comes in,
>> I've tried a number of different phones, both cordless and corded but it
>> makes no difference.
>>
>> Any suggestions
>
> You might need one of these, I did using a PAP2
>
> "RJ11 Adaptor with Ring Capacitor"
I ran into similar problems back in the day with the Tesco IPA1000
adapter. My fix was to plug an ADSL microfilter into it, and connect
the phone through it. If you have a spare one lying around, it's worth
a shot.
On 13/05/2011 18:35, Soruk wrote:
> On 2011-05-13, I'm Old Gregg<mantrid.a@ukonline.invalid> wrote:
>>
>>>
>>>> However the phone connected to it doesn't ring when a call comes in,
>>> I've tried a number of different phones, both cordless and corded but it
>>> makes no difference.
>>>
>>> Any suggestions
>>
>> You might need one of these, I did using a PAP2
>>
>> "RJ11 Adaptor with Ring Capacitor"
>
> I ran into similar problems back in the day with the Tesco IPA1000
> adapter. My fix was to plug an ADSL microfilter into it, and connect
> the phone through it. If you have a spare one lying around, it's worth
> a shot.
>
Even though I'm using a Virgin Media cable connection ?
On 13/05/2011 16:51, Roger Mills wrote:
> On 12/05/2011 19:59, Blu wrote:
>>
>> I have a Patton micro ATA.
>>
>> The configuration looks correct to the untrained eye, it says it is up
>> and running, shows my correct a/c number as the user name, its set to
>> sip.voipfone.net as suggested by their support with the right port
>> number, and it correctly tells me the number of messages waiting.
>>
>> It makes outgoing calls fine.
>>
>> However the phone connected to it doesn't ring when a call comes in,
>> I've tried a number of different phones, both cordless and corded but it
>> makes no difference.
>>
>> Any suggestions of things to try please
>>
>> Blu.
>>
>> Router is Linksys WRT54G
>
> What happens if you pick up[1] the phone when it should be ringing, but
> isn't - can you then hold a conversation with the calling party? Or do
> you just get dial-tone, indicating that the call hasn't been connected?
>
> [1] You could test this by ringing from a landline (or mobile) in the
> next room (say).
Voipfone support have told me the calls just "don't connect" and thus go
straight to voicemail. I'll try this though, thanks for the response.
In article <xwizp.5947$%o4.2047@newsfe16.ams2>, blu@green.com says...
>
> >
>
> Even though I'm using a Virgin Media cable connection ?
>
> Blu
It's use has nothing to do with how broadband is delivered to you. The
ADSL filter suggested contains a ring capacitor to provide the ring
signal on pin 3. He could have suggested a beter solution of using a
PBX master socket but that would be harder to find and probably cost
more.
I'm suprised that all the test phones you tried needed the pin-3 ringing
signal but, it you have a filter, it's worth a try,
"Blu" <blu@green.com> wrote in message news:xwizp.5947$%o4.2047@newsfe16.ams2...
> On 13/05/2011 18:35, Soruk wrote:
>> On 2011-05-13, I'm Old Gregg<mantrid.a@ukonline.invalid> wrote:
>>>
>>>>
>>>>> However the phone connected to it doesn't ring when a call comes in,
>>>> I've tried a number of different phones, both cordless and corded but it
>>>> makes no difference.
>>>>
>>>> Any suggestions
>>>
>>> You might need one of these, I did using a PAP2
>>>
>>> "RJ11 Adaptor with Ring Capacitor"
>>
>> I ran into similar problems back in the day with the Tesco IPA1000
>> adapter. My fix was to plug an ADSL microfilter into it, and connect
>> the phone through it. If you have a spare one lying around, it's worth
>> a shot.
>>
>
> Even though I'm using a Virgin Media cable connection ?
>
> Blu
An ADSL filter just happens to have the component that emulates A UK master
socket and couples the ringing current to pin 3 of the socket. Only a minority
of phones actually require this today.
What we all really need to know in order to take this further is the answer to the
following question.
We know the phone doesn't ring, but if you anticipate a call (because you made it yourself)
can you answer it and have a conversation?
"Blu" wrote in message news:Jyizp.5948$%o4.140@newsfe16.ams2...
On 13/05/2011 16:51, Roger Mills wrote:
> On 12/05/2011 19:59, Blu wrote:
>>
>> I have a Patton micro ATA.
>>
>> The configuration looks correct to the untrained eye, it says it is up
>> and running, shows my correct a/c number as the user name, its set to
>> sip.voipfone.net as suggested by their support with the right port
>> number, and it correctly tells me the number of messages waiting.
>>
>> It makes outgoing calls fine.
>>
>> However the phone connected to it doesn't ring when a call comes in,
>> I've tried a number of different phones, both cordless and corded but it
>> makes no difference.
>>
>> Any suggestions of things to try please
>>
>> Blu.
>>
>> Router is Linksys WRT54G
>
> What happens if you pick up[1] the phone when it should be ringing, but
> isn't - can you then hold a conversation with the calling party? Or do
> you just get dial-tone, indicating that the call hasn't been connected?
>
> [1] You could test this by ringing from a landline (or mobile) in the
> next room (say).
>Voipfone support have told me the calls just "don't connect" and thus go
>straight to voicemail. I'll try this though, thanks for the response.
If the calls aren't connecting, this wont help. Is there a parameter on your
ATA that shows the last number that called and the time and date? This is a
feature on most. If you check this when you try and call the number from
your mobile (or as others have suggested just pick the phone up when you
here the ringing tone on your mobile and see if it connects) you'll know if
the calls are getting as far as the ATA.
You may also be able to make calls without the ATA actually registering(I
dont know if this is the case with voipfone or not). If you log into
voipfone on the website is there anywhere that shows you what devices are
registered to your account? If it's registered but you still cant receive
calls then there is probably a port problem on your router. If the ATA is
correctly showing the messages waiting then I suspect its registering ok.
Voipfone say they dont support STUN but this can often help if the problem
is with ports. Try setting a stun server on your ATA - you can use
stun.sipgate.net (it doesnt matter that its a sipgate one and see if this
helps. If this doesn't work try forwarding port 5060 in your router to the
address of your ATA - if this makes the phone ring but you only get one way
audio you may then have to forward the RTP port ranges that are configured
in your ATA as well.
"Roger" wrote in message news:iyszp.145$Ky.69@newsfe24.ams2...
"Blu" wrote in message news:Jyizp.5948$%o4.140@newsfe16.ams2...
On 13/05/2011 16:51, Roger Mills wrote:
> On 12/05/2011 19:59, Blu wrote:
>>
>> I have a Patton micro ATA.
>>
>> The configuration looks correct to the untrained eye, it says it is up
>> and running, shows my correct a/c number as the user name, its set to
>> sip.voipfone.net as suggested by their support with the right port
>> number, and it correctly tells me the number of messages waiting.
>>
>> It makes outgoing calls fine.
>>
>> However the phone connected to it doesn't ring when a call comes in,
>> I've tried a number of different phones, both cordless and corded but it
>> makes no difference.
>>
>> Any suggestions of things to try please
>>
>> Blu.
>>
>> Router is Linksys WRT54G
>
> What happens if you pick up[1] the phone when it should be ringing, but
> isn't - can you then hold a conversation with the calling party? Or do
> you just get dial-tone, indicating that the call hasn't been connected?
>
> [1] You could test this by ringing from a landline (or mobile) in the
> next room (say).
>Voipfone support have told me the calls just "don't connect" and thus go
>straight to voicemail. I'll try this though, thanks for the response.
If the calls aren't connecting, this wont help. Is there a parameter on your
ATA that shows the last number that called and the time and date? This is a
feature on most. If you check this when you try and call the number from
your mobile (or as others have suggested just pick the phone up when you
here the ringing tone on your mobile and see if it connects) you'll know if
the calls are getting as far as the ATA.
You may also be able to make calls without the ATA actually registering(I
dont know if this is the case with voipfone or not). If you log into
voipfone on the website is there anywhere that shows you what devices are
registered to your account? If it's registered but you still cant receive
calls then there is probably a port problem on your router. If the ATA is
correctly showing the messages waiting then I suspect its registering ok.
Voipfone say they dont support STUN but this can often help if the problem
is with ports. Try setting a stun server on your ATA - you can use
stun.sipgate.net (it doesnt matter that its a sipgate one and see if this
helps. If this doesn't work try forwarding port 5060 in your router to the
address of your ATA - if this makes the phone ring but you only get one way
audio you may then have to forward the RTP port ranges that are configured
in your ATA as well.
Let us know how you get one.
I've just looked at the Patton website. It looks like NAT traversal is set
to "None" as default in the manual. This wont work behind a NAT router
(unless your ATA is plugged directly into an old style virgin cable modem
you'll be behind a NAT router). Before trying the STUN method I mentinoed
above , set the UPNP option in this setting and see if this cures your
problem. (Make sure UPNP is turned on on your router as well - normally it
is by default).
On 14/05/2011 09:26, John Weston wrote:
> In article<xwizp.5947$%o4.2047@newsfe16.ams2>, blu@green.com says...
>>
>
>>>
>>
>> Even though I'm using a Virgin Media cable connection ?
>>
>> Blu
>
> It's use has nothing to do with how broadband is delivered to you. The
> ADSL filter suggested contains a ring capacitor to provide the ring
> signal on pin 3. He could have suggested a beter solution of using a
> PBX master socket but that would be harder to find and probably cost
> more.
>
> I'm suprised that all the test phones you tried needed the pin-3 ringing
> signal but, it you have a filter, it's worth a try,
>
ahh - OK. I'll certainly give this a try as I have a number of filters
left over from my days as an unhappy customer of BT.
> If the calls aren't connecting, this wont help. Is there a parameter on
> your
> ATA that shows the last number that called and the time and date? This is a
> feature on most. If you check this when you try and call the number from
> your mobile (or as others have suggested just pick the phone up when you
> here the ringing tone on your mobile and see if it connects) you'll know if
> the calls are getting as far as the ATA.
I've tried calling myself from my mobile. I am unable to have a
conversation and with the handset connected to the ATA, all I get is the
dial tone.
I don't think the calls are connecting.
>
> You may also be able to make calls without the ATA actually registering(I
> dont know if this is the case with voipfone or not). If you log into
> voipfone on the website is there anywhere that shows you what devices are
> registered to your account? If it's registered but you still cant receive
> calls then there is probably a port problem on your router. If the ATA is
> correctly showing the messages waiting then I suspect its registering ok.
The ATA appears to register with voipfone with no problem, it shows as
registered and correctly gives my account number as my username,
although it usually took it a while to show the messages waiting. I've
made the change you suggested to Nat Traversal to uPNP and when I reboot
the ATA and go into its interface, it shows the messages waiting
immediately.
However it hasn't solved the problem of not connecting and calls going
straight to voicemail.
> cable modem you'll be behind a NAT router). Before trying the STUN
> method I mentinoed above , set the UPNP option in this setting and see
> if this cures your problem. (Make sure UPNP is turned on on your router
> as well - normally it is by default).
I've got setting the stun server on my 'to do' list. Thanks very much
for your help so far.
> An ADSL filter just happens to have the component that emulates A UK master
> socket and couples the ringing current to pin 3 of the socket. Only a minority
> of phones actually require this today.
>
> What we all really need to know in order to take this further is the answer to the
> following question.
> We know the phone doesn't ring, but if you anticipate a call (because you made it yourself)
> can you answer it and have a conversation?
Thanks for this Graham.
No, if I call my voip number from my mobile phone at no point am I able
to have a conversation, all I get from the Voip handset is the dial tone
no matter where in the process I pick the handset up.
I've made the change Roger suggested, setting NAT Traversal to uPNP.
When I reboot, the ATA interface shows the number of waiting messages
far more quickly than it did before, indeed straight away, but it hasn't
solved the issue of calls not connecting to the ATA.
> If the calls aren't connecting, this wont help. Is there a parameter on
> your
> ATA that shows the last number that called and the time and date? This is a
> feature on most. If you check this when you try and call the number from
> your mobile (or as others have suggested just pick the phone up when you
> here the ringing tone on your mobile and see if it connects) you'll know if
> the calls are getting as far as the ATA.
No parameter for last call that I can see, but voipfone have confirmed
to me that when they call the number their calls are not getting through
to my ATA.
> Voipfone say they dont support STUN but this can often help if the problem
> is with ports. Try setting a stun server on your ATA - you can use
> stun.sipgate.net (it doesnt matter that its a sipgate one and see if this
> helps.
I've tried the stun server and the proxy server nat.voipfone.co.uk but
neither has made a difference.
If this doesn't work try forwarding port 5060 in your router to the
> address of your ATA - if this makes the phone ring but you only get one way
> audio you may then have to forward the RTP port ranges that are configured
> in your ATA as well.
This is on my to do list, will let you know later.
>
> Let us know how you get one.
>
> I've just looked at the Patton website. It looks like NAT traversal is
> set to "None" as default in the manual. This wont work behind a NAT
> router (unless your ATA is plugged directly into an old style virgin
> cable modem you'll be behind a NAT router). Before trying the STUN
> method I mentinoed above , set the UPNP option in this setting and see
> if this cures your problem. (Make sure UPNP is turned on on your router
> as well - normally it is by default).
As I said yesterday, the NAT traversal change made for a more robust
connection to voipfone, but hasn't solved the problem. By robust I mean
the display of the number of messages appears immediately whilst before
it might take 30 mins or so to appear. This robustness also happen with
both the stun server and the outbound proxy settings.
Blu
firstly, always obscure or remove your ip address and account info from
informatoin you post in messages.
It looks like your router is blocking the requests from voipfone. Forward
port 5060 as suggested and see if that resolves it (you'll also then need to
set up a fixed range of RTP ports on the ATA and forward those as well).
On May 16, 11:36*am, Blu <b...@green.com> wrote:
> Does all this stuff make any sense to anyone here ?
>
> We send multiple INVITE requests to the device, asking it to accept the
> inbound call.
>
> We get nothing back at all, So give up and pass the call to voicemail.
>
> Trace of this below:
>
> match: 30137731
>
> U 2011/05/16 10:17:37.514296 195.189.173.205:5060 -> 195.189.173.10:5060
> INVITE *sip:30137...@192.168.1.101:5060 SIP/2.0.
> Via: SIP/2.0/UDP 195.189.173.205:5060;branch=z9hG4bK1aace573;rport.
> From: "anonymous" <sip:2131665...@195.189.173.205>;tag=as2d7614b7.
> To: < sip:30137...@192.168.1.101:5060>.
> Contact: <sip:2131665...@195.189.173.205>.
> Call-ID: 28f66c097d0e49703c5b182139d22...@195.189.173.205.
> CSeq: 102 INVITE.
> User-Agent: Voipfone Sip Network.
> Date: Mon, 16 May 2011 09:19:47 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
> Content-Type: application/sdp.
> Content-Length: 349.
> .
> v=0.
> o=root 2094 2094 IN IP4 195.189.173.205.
> s=session.
> c=IN IP4 195.189.173.205.
> t=0 0.
> m=audio 27216 RTP/AVP 8 0 2 97 3 110 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:2 G726-32/8000.
> a=rtpmap:97 iLBC/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:110 speex/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
>
> U 2011/05/16 10:17:37.914213 195.189.173.12:5065 -> 195.189.173.10:5060
> REGISTER sip:voipfone.co.uk SIP/2.0.
> Record-Route: <sip:195.189.173.12:5065;ftag=K59f2-mgYqF;lr=on>.
> From: sip:30137...@voipfone.co.uk;tag=K59f2-mgYqF.
> To: sip:30137...@voipfone.co.uk.
> Call-ID: uXyiV0-lYOa...@voipfone.co.uk.
> CSeq: 27 REGISTER.
> Via: SIP/2.0/UDP 195.189.173.12:5065;branch=0.
> Via: SIP/2.0/UDP
> 192.168.1.101:5060;rport=5060;received=94.174.21.2 26;branch=z9hG4bKeJ0f2-p3 8Gh1vM0.
> Contact: sip:30137...@94.174.21.226:5060.
> Max-Forwards: 16.
> Route: <sip:195.189.173.12:5065>.
> Authorization: Digest
> username="30137731",realm="asterisk",uri="sip:voip fone.co.uk",response="924 1e32e6a672a3b4555d0febb741f61",nonce="vonyarda".
> User-Agent: Patton Smartlink MATA <4.01.001 OE EN MA (0309)><00a0ba02fa7a>.
> Expires: 60.
> Content-Length: 0.
> X-VNRI: 94.174.21.226:5060.
> .
>
> U 2011/05/16 10:17:37.915602 195.189.173.10:5060 -> 195.189.173.137:5062
> REGISTER sip:voipfone.co.uk SIP/2.0.
> X-Voipfone-Real-Ip: 195.189.173.12:5065.
> Via: SIP/2.0/UDP 195.189.173.10:5060;branch=z9hG4bK-324c7782.
> Via: SIP/2.0/UDP
> 195.189.173.12:5065;branch=0;received=195.189.173. 12;rport=5065.
> Record-Route: <sip:195.189.173.10:5060;lr=on>.
> Record-Route: <sip:195.189.173.12:5065;ftag=K59f2-mgYqF;lr=on>.
> From: sip:30137...@voipfone.co.uk;tag=K59f2-mgYqF.
> To: sip:30137...@voipfone.co.uk.
> Call-ID: uXyiV0-lYOa...@voipfone.co.uk.
> CSeq: 27 REGISTER.
> Via: SIP/2.0/UDP
> 192.168.1.101:5060;rport=5060;received=94.174.21.2 26;branch=z9hG4bKeJ0f2-p3 8Gh1vM0.
> Contact: sip:30137...@94.174.21.226:5060.
> Max-Forwards: 16.
> Route: <sip:195.189.173.12:5065>.
> Authorization: Digest
> username="30137731",realm="asterisk",uri="sip:voip fone.co.uk",response="924 1e32e6a672a3b4555d0febb741f61",nonce="vonyarda".
> User-Agent: Patton Smartlink MATA <4.01.001 OE EN MA (0309)><00a0ba02fa7a>.
> Expires: 60.
> Content-Length: 0.
> X-VNRI: 94.174.21.226:5060.
> .
>
> U 2011/05/16 10:17:37.917793 195.189.173.10:5060 -> 195.189.173.12:5065
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 195.189.173.12:5065;branch=0;received=195.189.173. 12;rport=5065.
> Record-Route: <sip:195.189.173.10:5060;lr=on>.
> Record-Route: <sip:195.189.173.12:5065;ftag=K59f2-mgYqF;lr=on>.
> From: sip:30137...@voipfone.co.uk;tag=K59f2-mgYqF.
> To: sip:30137...@voipfone.co.uk.
> Call-ID: uXyiV0-lYOa...@voipfone.co.uk.
> CSeq: 27 REGISTER.
> Via: SIP/2.0/UDP
> 192.168.1.101:5060;rport=5060;received=94.174.21.2 26;branch=z9hG4bKeJ0f2-p3 8Gh1vM0.
> From: sip:30137...@voipfone.co.uk;tag=K59f2-mgYqF.
> To: sip:30137...@voipfone.co.uk.
> Call-ID: uXyiV0-lYOa...@voipfone.co.uk.
> CSeq: 27 REGISTER.
> Contact: sip:30137...@94.174.21.226:5060;expires=60.
> Expires: 60.
> Date: Mon, 16 May 2011 09:19:47 GMT.
> Min-Expires: 60.
> User-Agent: Voipfone.
> Content-Length: 0.
> .
FYI I spoke to one of the customer service girls at Voipfone today as
I was having some Nat and audio issues, and they informed me that the
sip.voipfone.co.uk is going to be retired in the near future, and that
I should register my phones to sip.voipfone.net I'm not quite sure
what the difference is between the new proxy and the old one is, but
it solved all my nat and audio issues, I also no longer needed to use
their nat proxy.
May be this will help with your issues too worth a try, I guess
On 16/05/2011 20:33, Roger wrote:
> Blu
> firstly, always obscure or remove your ip address and account info from
> informatoin you post in messages.
>
> It looks like your router is blocking the requests from voipfone.
> Forward port 5060 as suggested and see if that resolves it (you'll also
> then need to set up a fixed range of RTP ports on the ATA and forward
> those as well).
>
> Roger.
Roger,
I normally would obscure personal details, but I really didn't
understand that much about what I was sending. Fortunately I don't have
a fixed ip address, so I can change that at least.
Port Forwarding allows me to choose TCP, UDP or both. For now I'm
forwarding both, is this correct ?
Under SIP Parameters there is a section called RTP Parameters, with a
setting for RTP Port Min and for RTP Port Max. Any suggestions for the
range to use. I don't want to clash with something else I use.
Then I guess I set the same range in the router, again should this be
TCP, UDP or both.
On 17/05/2011 22:15, Blu wrote:
> On 16/05/2011 20:33, Roger wrote:
>> Blu
>> firstly, always obscure or remove your ip address and account info from
>> informatoin you post in messages.
>>
>> It looks like your router is blocking the requests from voipfone.
>> Forward port 5060 as suggested and see if that resolves it (you'll also
>> then need to set up a fixed range of RTP ports on the ATA and forward
>> those as well).
>>
>> Roger.
>
> Roger,
>
> I normally would obscure personal details, but I really didn't
> understand that much about what I was sending. Fortunately I don't have
> a fixed ip address, so I can change that at least.
>
> Port Forwarding allows me to choose TCP, UDP or both. For now I'm
> forwarding both, is this correct ?
>
> Under SIP Parameters there is a section called RTP Parameters, with a
> setting for RTP Port Min and for RTP Port Max. Any suggestions for the
> range to use. I don't want to clash with something else I use.
>
> Then I guess I set the same range in the router, again should this be
> TCP, UDP or both.
>
> Thanks for your help sp far.
>
> Blu.
>
Well, its set to TCP and UDP and for the RTP packets I've forwarded the
range from the TCP/UDP port table which is 16,000 or so to 34,000 or so.
Turned off both the router and the ATA, turned the router back on 30
mins later and then the ATA. Saved changes of course.
Still does not work, calls go straight to voicemail. I think I'll try
and get a refund off broadbandbuyer although they may insist on sending
me a replacement ATA, I doubt you can actually repair these things.
Seems to me VOIP and ATAs etc are only really for telephone engineering
types, I see now wny people use Skype. Lets just hope M$ dont bugger
that up for people.
On 16/05/2011 22:14, Tmorley wrote:
> On May 16, 11:36 am, Blu<b...@green.com> wrote:
>> Does all this stuff make any sense to anyone here ?
>>
>> We send multiple INVITE requests to the device, asking it to accept the
>> inbound call.
>>
>> We get nothing back at all, So give up and pass the call to voicemail.
>>
>> Trace of this below:
>>
>> >
> FYI I spoke to one of the customer service girls at Voipfone today as
> I was having some Nat and audio issues, and they informed me that the
> sip.voipfone.co.uk is going to be retired in the near future, and that
> I should register my phones to sip.voipfone.net I'm not quite sure
> what the difference is between the new proxy and the old one is, but
> it solved all my nat and audio issues, I also no longer needed to use
> their nat proxy.
>
> May be this will help with your issues too worth a try, I guess
>
> T
Thanks for your contribution. The only time the ATA actually worked was
when it was set to sip.voipfone.co.uk but it used to change its status
from 'Registered' to 'Error Forbidden' after about 3 days uptime.
On Voipfone's advice I changed it to sip.voipfone.net and low and behold
it stopped ringing at all and changing back to sip.voipfone.co.uk didn't
help.
I think its time to get a refund or replacement ATA, and for now its
back to softphones and Skype !!
"Blu" wrote in message news:8VCAp.4405$yI6.1892@newsfe06.ams2...
On 17/05/2011 22:15, Blu wrote:
> On 16/05/2011 20:33, Roger wrote:
>> Blu
>> firstly, always obscure or remove your ip address and account info from
>> informatoin you post in messages.
>>
>> It looks like your router is blocking the requests from voipfone.
>> Forward port 5060 as suggested and see if that resolves it (you'll also
>> then need to set up a fixed range of RTP ports on the ATA and forward
>> those as well).
>>
>> Roger.
>
> Roger,
>
> I normally would obscure personal details, but I really didn't
> understand that much about what I was sending. Fortunately I don't have
> a fixed ip address, so I can change that at least.
>
> Port Forwarding allows me to choose TCP, UDP or both. For now I'm
> forwarding both, is this correct ?
>
> Under SIP Parameters there is a section called RTP Parameters, with a
> setting for RTP Port Min and for RTP Port Max. Any suggestions for the
> range to use. I don't want to clash with something else I use.
>
> Then I guess I set the same range in the router, again should this be
> TCP, UDP or both.
>
> Thanks for your help sp far.
>
> Blu.
>
Well, its set to TCP and UDP and for the RTP packets I've forwarded the
range from the TCP/UDP port table which is 16,000 or so to 34,000 or so.
Turned off both the router and the ATA, turned the router back on 30
mins later and then the ATA. Saved changes of course.
Still does not work, calls go straight to voicemail. I think I'll try
and get a refund off broadbandbuyer although they may insist on sending
me a replacement ATA, I doubt you can actually repair these things.
Seems to me VOIP and ATAs etc are only really for telephone engineering
types, I see now wny people use Skype. Lets just hope M$ dont bugger
that up for people.
Thanks for all your help Roger and Graham.
A very frustrated Blu.
(time for a large brandy then bed !!)
Just before you send it back - why not set up a free sipgate account and see
if that works - then you'll now if its the ATA or something to do with
voipfone.
>>
>
> Well, its set to TCP and UDP and for the RTP packets I've forwarded the
> range from the TCP/UDP port table which is 16,000 or so to 34,000 or so.
>
> Turned off both the router and the ATA, turned the router back on 30
> mins later and then the ATA. Saved changes of course.
>
> Still does not work, calls go straight to voicemail. I think I'll try
> and get a refund off broadbandbuyer although they may insist on sending
> me a replacement ATA, I doubt you can actually repair these things.
>
> Seems to me VOIP and ATAs etc are only really for telephone engineering
> types, I see now wny people use Skype. Lets just hope M$ dont bugger
> that up for people.
>
> Thanks for all your help Roger and Graham.
>
> A very frustrated Blu.
>
> (time for a large brandy then bed !!)
>
>
> Just before you send it back - why not set up a free sipgate account and see if that works - then you'll now if its the ATA or
> something to do with voipfone.
Yes, I was going to suggest that too. Also, you could download the x-lite softphone
and see if that has the same issues as the hardware.
On 19/05/2011 23:31, Graham. wrote:
>>>
>>
>> Well, its set to TCP and UDP and for the RTP packets I've forwarded the
>> range from the TCP/UDP port table which is 16,000 or so to 34,000 or so.
>>
>> Turned off both the router and the ATA, turned the router back on 30
>> mins later and then the ATA. Saved changes of course.
>>
>> Still does not work, calls go straight to voicemail. I think I'll try
>> and get a refund off broadbandbuyer although they may insist on sending
>> me a replacement ATA, I doubt you can actually repair these things.
>>
>> Seems to me VOIP and ATAs etc are only really for telephone engineering
>> types, I see now wny people use Skype. Lets just hope M$ dont bugger
>> that up for people.
>>
>> Thanks for all your help Roger and Graham.
>>
>> A very frustrated Blu.
>>
>> (time for a large brandy then bed !!)
>>
>>
>> Just before you send it back - why not set up a free sipgate account and see if that works - then you'll now if its the ATA or
>> something to do with voipfone.
>
> Yes, I was going to suggest that too. Also, you could download the x-lite softphone
> and see if that has the same issues as the hardware.
>
I have XLite on the Mac and Zoiper on the Windows 7 PC. Both work
perfectly apart from the issue that the computer has to be on !!
Haven't used either since making the changes to the router settings to
forward port 5060 and the rtp packets though, guess I'd have to disable
those changes.
Does this suggest that the problem is with the ATA ?
Trying Sipgate is a job for the weekend most likely.
"Blu" <blu@green.com> wrote in message news:R6Wyp.56$ZQ2.9@newsfe11.ams2...
>
> I have a Patton micro ATA.
>
> The configuration looks correct to the untrained eye, it says it is up and
> running, shows my correct a/c number as the user name, its set to
> sip.voipfone.net as suggested by their support with the right port number,
> and it correctly tells me the number of messages waiting.
>
> It makes outgoing calls fine.
>
> However the phone connected to it doesn't ring when a call comes in, I've
> tried a number of different phones, both cordless and corded but it makes
> no difference.
>
> Any suggestions of things to try please
>
> Blu.
>
> Router is Linksys WRT54G
Hi,
A few thoughts. Some commonalities which may be useful too.
1) Factory Reset - A complete factory reset (should be a button on the
device) would be recommended or at least a good idea. The device may have
conflicting settings or acquired a hangover cross-setting or corruption.
2) Router / SIP Systemics - Their are commonalities across
implementations (or some might [perhaps] [overly] generously say
mis-implementations) in or of routers and SIP devices / software[s] (and as
well, say more sensibly, "alternate" or "simple" products / services
[devices, software[s], etc.]). Typically these manifest very loosely,
briefly and total novice (style) as:
- Routers: hang-ups or ("loose"[!]) implementations - impacting or
affecting, port forwarding, SIP assistance / handling protocols or packages
(e.g. ALGs - Application Layer Gateways), DMZ functionality, trigger -
detect - latch - release on or with NAT (as an example), etc.
- SIP devices / software: partial or varying interpretation
implementations (and as / similar to router entries [above]) - impacting or
affecting, configuration (differing terms or use of terms or missing terms,
lack of options [or an option], etc.), one way audio / no incoming call
detect (firewall handling / traversal issues, NAT / router, etc.),
flexibility (much as or resultant to previous - e.g. some functions or
features not working / partly implemented or unavailable or only available
to specific configurations, devices [or much more infrequently and usually
only special case scenarios - only {available} to devices or configurations
with special or specific "additional" equipment, options, etc. {e.g. " super
call waiting PLUS " by " XYZ Manufacturing " and only available on their
devices, systems and configurations, etc., etc.]).
- Generic - Much as the previous (ones) - corruption / data [/]
stack issues, config issues or limitations (or exceptionally additions /
extras), implementation issues or limitations, fade (losing config or
environment / parameter variables over the medium term) ([can be]
independent of the device or its function / role / duty, etc.).
3) Service Provider(s) - Your ISP (internet service provider), your VSP
(voice service provider ? ), etc. One or more of these may have issues,
configuration problems or limitations (additions), etc., service
restrictions / incompatibilites, etc., etc.
Hence:
- consider a factory reset of your device(s), and a careful slow
("manual" - if you are up to it) configuration ("re-configuration") (perhaps
write down any settings you need before commencing - as long as you are sure
they are not corruption / cross- settings, etc. - e.g. "known corrects" such
as perhaps your ISP username, etc. [ - again, if you need it, and bearing in
mind that you should have the originals or the originals for these
somewhere, and should be known or know them to cross-reference against to be
sure what you are storing is the original or correct version and not corrupt
or other, etc.])
- evaluate your service provider(s) - I would particularly focus on your
ISP here (potentially) - e.g. some block VoIP or other ports, some implement
("multi-layer") NAT(s) (which can render a network partially or totally
unnavigable).
- SIP devices / software - some of these have varying configuration
approaches and complexity. Hence, approaching the task from a perhaps
"easier" direction, a proportion will assume certain settings where others
will either not permit them or not have / work at all where yet others will
require a very complete or full technical (or complex) configuration.
- router[s] ( / "perhaps as distinct from routing") - common approaches
for users facing issues, include, DMZ (try this - temporarily / testing only
[with appropriate, matching, destinations / equipment - i.e. a device or
computer that can face, cope with or is configured for "direct" internet
access - i.e. one ready for a testing or full access challenging
environment]), port forwarding (often not too many ports or ranges or - ...
gulp, router, strange / mis-behaviour), (SIP) ALGs (or equivalent[s]) try on
and off, enhanced or extra features off (UPnP, etc.) (or on?), always good /
standard advice, or good idea or sensible to use wired connections for
development or testing (new equipment, new gear, etc.), ("forced") (basic)
configuration for WAN side (xDSL settings, etc. - but that is getting a bit
carried away / advanced). (lol - ... [advanced / joke ?] don't forget
matching settings in your SIP client).
(Don't forget, the foregoing may require trying / "switching" independently,
sometimes including or requiring a factory reset between [but that is
getting a bit keen / carried away - sorry, "detailed pages" "here", e.g.
sometimes you can second guess combinations that are likely to require such
drasticness]), or to be used in combination[s]).
So, to avoid getting carried away, perhaps for yourself or your own
situation / environment, try: factory reset, (basic) check of your service
provider(s) ( ~ ISP) (mobile broadband environments are particularly prone
to not pass [some or all] VoIP traffic, ULTRA bargain or special ISP
packages / services, etc.) and simple check of your ATA (VoIP configuration)
(key aspects, again perhaps if you are keen [enough], try combinations /
variations - SIP port ranges, SIP server / hostnames, network / firewall -
traversal or identification, configuration features or tools), simple check
of your (home) router (key aspects, again consider combos / variations, DMZ,
SIP help / special settings [if any], special or compatibility settings,
port forwarding, etc.).
If that doesn't work (sorry for skipping specifics / [ultra] details on some
of the foregoing), most probable, equipment incompatibility ( / too or few
configurability).... Some vendors hardware, set-ups / configurations, work
poorly or not at all together, where some other (combinations) work
immediately and / or excellently straight out of the box.
Don't forget line impedance (as for or if available with your ATA) (i.e.
match your telephone - UK setting?).
Hope that helps. One simpler potential tip for you is, request people to
advise examples of known simple or good hardware (and / or combinations), or
look at (the same /) guidance on subject related user communications (feeds)
(usergroups, etc.).
"News Reader" <some@where.invalid> wrote in message
news:irec9g$edj$1@speranza.aioe.org...
>
> "Blu" <blu@green.com> wrote in message
> news:R6Wyp.56$ZQ2.9@newsfe11.ams2...
>>
>> I have a Patton micro ATA.
>>
>> The configuration looks correct to the untrained eye, it says it is up
>> and running, shows my correct a/c number as the user name, its set to
>> sip.voipfone.net as suggested by their support with the right port
>> number, and it correctly tells me the number of messages waiting.
>>
>> It makes outgoing calls fine.
>>
>> However the phone connected to it doesn't ring when a call comes in, I've
>> tried a number of different phones, both cordless and corded but it makes
>> no difference.
>>
>> Any suggestions of things to try please
>>
>> Blu.
>>
>> Router is Linksys WRT54G
>
>
> Hi,
>
>
> A few thoughts. Some
If you might want some more particular help on the port forwarding (router
type) issues. You could try the link (or the post itself [ - lol - keenness
: - ) ]) in my previous post (below - start half way down the page - where
the indentation marks end [ the " > " marks / indicators]):
On 23/05/2011 20:24, News Reader wrote:
> "Blu"<blu@green.com> wrote in message news:R6Wyp.56$ZQ2.9@newsfe11.ams2...
>>
>> I have a Patton micro ATA.
>>
>> The configuration looks correct to the untrained eye, it says it is up and
>> running, shows my correct a/c number as the user name, its set to
>> sip.voipfone.net as suggested by their support with the right port number,
>> and it correctly tells me the number of messages waiting.
>>
>> It makes outgoing calls fine.
>>
>> However the phone connected to it doesn't ring when a call comes in, I've
>> tried a number of different phones, both cordless and corded but it makes
>> no difference.
>>
>> Any suggestions of things to try please
>>
>> Blu.
>>
>> Router is Linksys WRT54G
>
>
> Hi,
>
>
> A few thoughts. Some commonalities which may be useful too.
>
> 1) Factory Reset - A complete factory reset (should be a button on the
> device) would be recommended or at least a good idea. The device may have
> conflicting settings or acquired a hangover cross-setting or corruption.
>
> 2) Router / SIP Systemics - Their are commonalities across
> implementations (or some might [perhaps] [overly] generously say
> mis-implementations) in or of routers and SIP devices / software[s] (and as
> well, say more sensibly, "alternate" or "simple" products / services
> [devices, software[s], etc.]). Typically these manifest very loosely,
> briefly and total novice (style) as:
>
> - Routers: hang-ups or ("loose"[!]) implementations - impacting or
> affecting, port forwarding, SIP assistance / handling protocols or packages
> (e.g. ALGs - Application Layer Gateways), DMZ functionality, trigger -
> detect - latch - release on or with NAT (as an example), etc.
>
> - SIP devices / software: partial or varying interpretation
> implementations (and as / similar to router entries [above]) - impacting or
> affecting, configuration (differing terms or use of terms or missing terms,
> lack of options [or an option], etc.), one way audio / no incoming call
> detect (firewall handling / traversal issues, NAT / router, etc.),
> flexibility (much as or resultant to previous - e.g. some functions or
> features not working / partly implemented or unavailable or only available
> to specific configurations, devices [or much more infrequently and usually
> only special case scenarios - only {available} to devices or configurations
> with special or specific "additional" equipment, options, etc. {e.g. " super
> call waiting PLUS " by " XYZ Manufacturing " and only available on their
> devices, systems and configurations, etc., etc.]).
>
> - Generic - Much as the previous (ones) - corruption / data [/]
> stack issues, config issues or limitations (or exceptionally additions /
> extras), implementation issues or limitations, fade (losing config or
> environment / parameter variables over the medium term) ([can be]
> independent of the device or its function / role / duty, etc.).
>
> 3) Service Provider(s) - Your ISP (internet service provider), your VSP
> (voice service provider ? ), etc. One or more of these may have issues,
> configuration problems or limitations (additions), etc., service
> restrictions / incompatibilites, etc., etc.
>
>
> Hence:
>
> - consider a factory reset of your device(s), and a careful slow
> ("manual" - if you are up to it) configuration ("re-configuration") (perhaps
> write down any settings you need before commencing - as long as you are sure
> they are not corruption / cross- settings, etc. - e.g. "known corrects" such
> as perhaps your ISP username, etc. [ - again, if you need it, and bearing in
> mind that you should have the originals or the originals for these
> somewhere, and should be known or know them to cross-reference against to be
> sure what you are storing is the original or correct version and not corrupt
> or other, etc.])
>
> - evaluate your service provider(s) - I would particularly focus on your
> ISP here (potentially) - e.g. some block VoIP or other ports, some implement
> ("multi-layer") NAT(s) (which can render a network partially or totally
> unnavigable).
>
> - SIP devices / software - some of these have varying configuration
> approaches and complexity. Hence, approaching the task from a perhaps
> "easier" direction, a proportion will assume certain settings where others
> will either not permit them or not have / work at all where yet others will
> require a very complete or full technical (or complex) configuration.
>
> - router[s] ( / "perhaps as distinct from routing") - common approaches
> for users facing issues, include, DMZ (try this - temporarily / testing only
> [with appropriate, matching, destinations / equipment - i.e. a device or
> computer that can face, cope with or is configured for "direct" internet
> access - i.e. one ready for a testing or full access challenging
> environment]), port forwarding (often not too many ports or ranges or - ...
> gulp, router, strange / mis-behaviour), (SIP) ALGs (or equivalent[s]) try on
> and off, enhanced or extra features off (UPnP, etc.) (or on?), always good /
> standard advice, or good idea or sensible to use wired connections for
> development or testing (new equipment, new gear, etc.), ("forced") (basic)
> configuration for WAN side (xDSL settings, etc. - but that is getting a bit
> carried away / advanced). (lol - ... [advanced / joke ?] don't forget
> matching settings in your SIP client).
>
> (Don't forget, the foregoing may require trying / "switching" independently,
> sometimes including or requiring a factory reset between [but that is
> getting a bit keen / carried away - sorry, "detailed pages" "here", e.g.
> sometimes you can second guess combinations that are likely to require such
> drasticness]), or to be used in combination[s]).
>
> So, to avoid getting carried away, perhaps for yourself or your own
> situation / environment, try: factory reset, (basic) check of your service
> provider(s) ( ~ ISP) (mobile broadband environments are particularly prone
> to not pass [some or all] VoIP traffic, ULTRA bargain or special ISP
> packages / services, etc.) and simple check of your ATA (VoIP configuration)
> (key aspects, again perhaps if you are keen [enough], try combinations /
> variations - SIP port ranges, SIP server / hostnames, network / firewall -
> traversal or identification, configuration features or tools), simple check
> of your (home) router (key aspects, again consider combos / variations, DMZ,
> SIP help / special settings [if any], special or compatibility settings,
> port forwarding, etc.).
>
> If that doesn't work (sorry for skipping specifics / [ultra] details on some
> of the foregoing), most probable, equipment incompatibility ( / too or few
> configurability).... Some vendors hardware, set-ups / configurations, work
> poorly or not at all together, where some other (combinations) work
> immediately and / or excellently straight out of the box.
>
> Don't forget line impedance (as for or if available with your ATA) (i.e.
> match your telephone - UK setting?).
>
> Hope that helps. One simpler potential tip for you is, request people to
> advise examples of known simple or good hardware (and / or combinations), or
> look at (the same /) guidance on subject related user communications (feeds)
> (usergroups, etc.).
>
>
> Best wishes,
>
>
>
>
> News Reader
>
>
>
Thanks for this. I'm no stranger to doing factory default resets and I
know the information required for both Voipfone and Sipgate now.
I got a new Windows 7 Sony Vaio which decided it would connect to all my
neighbours wireless networks but didn't want to play with mine. A
default factory reset of my router was one of the things that solved
that problem.
I'll do a reset and manual reconfigure tomorrow if I have the time.
ATA issues on voipfone 
Linear Mode
