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VOIP Beta Trial Feedback

acr
Rising Star
Posts: 217
Thanks: 11
Registered: 06-04-2007

VOIP Beta Trial Feedback

I am using a Draytek 2800VG router and have noticed that registration seems to be happening every 25 seconds even though the expiry time is set at 1 hour. I also have a connection with Sipgate and this registers correctly every hour.
Calls to/from mobiles/landlines are working very well with good call quality however I am unable to connect to my Sipgate number (using the **777 prefix) from Plusnet or my Plusnet number (using the **472 prefix) from Sipgate. In both cases I get an engaged tone. I can connect to/from both connections using their respective 01 numbers.
Recording a voice prompt also does not appear to be working. The recorded message received when dialing 188 from the VOIP phone asks for the 9 digit Pin Code previously provided on the web site. Whenever I enter this 9 digit pin the number is not accepted.
My settings for the router are:-
Register via: Auto
SIP Port: 5060
Domain/Realm: sip2.plus.net
Proxy: nat.plus.net:5082
Act as Outbound Proxy: Ticked
Display Name: <my name>
Account Number/Name: <SIP ID>
Authentication ID: unticked
Password: <my password>
Expiry Time: 1 hour
NAT Traversal Support: None

Edit:-
When dialling Plusnet/Sipgate (using **777/**472) the router's log shows error '[404] User Cannot be found at this address'. Have tried ticking/unticking the 'Act as Outbound Proxy' setting and ticking/unticking the 'Authentication ID' setting with the same results.
19 REPLIES
Ianwild
Grafter
Posts: 3,835
Registered: 05-04-2007

Re: VOIP Beta Trial Feedback

Maybe I'm bring Dim - Where did 188 come from? I don't recollect seeing that one on our trial (Note - Not all of the same Gradwell features will work with PlusNet, as you don't have quite the same product!)... Maybe it should work, but I can't see where a 9 digit PIN is, and direct acces this isn't a feature we requested.
To access voicemail, I'm using *1 from my Voip line, or dialling 01225 800899 from my mobile when out and about. Recording my own message worked fine in the voicemail menu.
I don't understand the 25 seconds registering thing - We'll get to the bottom of it though I'm sure.
Ian
acr
Rising Star
Posts: 217
Thanks: 11
Registered: 06-04-2007

Re: VOIP Beta Trial Feedback

Quote from: Ian
Where did 188 come from?

From the VOIP Trial Services menu (http://voip.plus.net) , I selected 'Voice Prompts' and then 'Record New Voice Prompt'. After entering a description for the prompt and clicking on 'continue' you get a nine digit Pin and a message to dial 188 from your VOIP phone to record the new voice prompt.
I am assuming that as it is part of the Plusnet VOIP Trial Services menu that the option should work and that it should allow me to record my own message.
Dialling *1 allows me to access voicemail but I can't find an option to record my own message.
puddy
Grafter
Posts: 1,571
Registered: 10-06-2007

Re: VOIP Beta Trial Feedback

I cannot dial my sipgate account either
plusnet to sipgate fail with code 404
Puddy
speedtouch 716wl
siemens gigaset 460ip phone
Ianwild
Grafter
Posts: 3,835
Registered: 05-04-2007

Re: VOIP Beta Trial Feedback

Quote from: ruslyn

From the VOIP Trial Services menu (http://voip.plus.net) , I selected 'Voice Prompts' and then 'Record New Voice Prompt'. After entering a description for the prompt and clicking on 'continue' you get a nine digit Pin and a message to dial 188 from your VOIP phone to record the new voice prompt.
Dialling *1 allows me to access voicemail but I can't find an option to record my own message.

Ah, OK, I see now - I learn something every day!
I just pressed *1 then 0 and got the options to record my voicemail messages.
Ian
Community Gaffer
Community Gaffer
Posts: 12,808
Thanks: 636
Fixes: 62
Registered: 04-04-2007

Re: VOIP Beta Trial Feedback

Quote from: ruslyn
Quote from: Ian
Where did 188 come from?

From the VOIP Trial Services menu (http://voip.plus.net) , I selected 'Voice Prompts' and then 'Record New Voice Prompt'. After entering a description for the prompt and clicking on 'continue' you get a nine digit Pin and a message to dial 188 from your VOIP phone to record the new voice prompt.
I am assuming that as it is part of the Plusnet VOIP Trial Services menu that the option should work and that it should allow me to record my own message.

Yes it is part of the trial, and yes it should work.
I set my own personal message up doing just this, although I used the geographical number (0870 861 6662) from a PSTN line to record the greeting.
I've just tried setting up another so I could give the 188 number a go and it seems to be working fine for me using X-Lite on the internal network.
Quote from: ruslyn
Calls to/from mobiles/landlines are working very well with good call quality however I am unable to connect to my Sipgate number (using the **777 prefix) from Plusnet or my Plusnet number (using the **472 prefix) from Sipgate. In both cases I get an engaged tone. I can connect to/from both connections using their respective 01 numbers.

I think SIP peering may be broke.
I logged into my BETA account on the internal network and was unable to call my PlusTalk SIP ID that my ATA is registered with back at home (**768 followed by my PlusTalk SIP ID).
I then logged into PlusTalk from another machine on the internal network and tried dialling from PlusTalk to Gradwell (***472 followed by my Gradwell SIP ID) - this didn't work either.
I tried using the domain naming convention too (eg. sip:123456@sip.plus.net) and that also fails.
Gradwell to Gradwell SIP calls are working fine.

Bob Pullen
Plusnet Products Team
If I've been helpful then please give thanks ⤵

jakejohnson
Dabbler
Posts: 22
Registered: 06-11-2007

Re: VOIP Beta Trial Feedback

Quote from: ruslyn
I am using a Draytek 2800VG router and have noticed that registration seems to be happening every 25 seconds even though the expiry time is set at 1 hour. I also have a connection with Sipgate and this registers correctly every hour.
Calls to/from mobiles/landlines are working very well with good call quality however I am unable to connect to my Sipgate number (using the **777 prefix) from Plusnet or my Plusnet number (using the **472 prefix) from Sipgate. In both cases I get an engaged tone. I can connect to/from both connections using their respective 01 numbers.
Recording a voice prompt also does not appear to be working. The recorded message received when dialing 188 from the VOIP phone asks for the 9 digit Pin Code previously provided on the web site. Whenever I enter this 9 digit pin the number is not accepted.
My settings for the router are:-
Register via: Auto
SIP Port: 5060
Domain/Realm: sip2.plus.net
Proxy: nat.plus.net:5082
Act as Outbound Proxy: Ticked
Display Name: <my name>
Account Number/Name: <SIP ID>
Authentication ID: unticked
Password: <my password>
Expiry Time: 1 hour
NAT Traversal Support: None

Edit:-
When dialling Plusnet/Sipgate (using **777/**472) the router's log shows error '[404] User Cannot be found at this address'. Have tried ticking/unticking the 'Act as Outbound Proxy' setting and ticking/unticking the 'Authentication ID' setting with the same results.

Hi ruslyn,
If it does not recognise the 9 digit pin and it is the correct pin then the problem may be the DTMF settings on the phone.  What DTMF options does the Draytek give you?
Thanks,
Jake
acr
Rising Star
Posts: 217
Thanks: 11
Registered: 06-04-2007

Re: VOIP Beta Trial Feedback

Quote from: Jake

What DTMF options does the Draytek give you?

The Draytek gives four options for the DTMF mode of the VOIP port - Inband, Outband (RFC2833), SIP INFO (cisco format) and SIP INFO (nortel format).  I was using the default option of 'Inbound' but by changing it to 'SIP INFO (cisco format)' the PIN is now being accepted and I have been able to record the voice prompt.
In case anybody else looking at this is also having the same problem with a Draytek 2800VG, or similar,  then you will need to change the DTMF mode by going to 'Phone Settings' under the VOIP menu. Then click on the port number that you are using followed by the 'Advanced' tab. The DTMF options are then shown.
That just leaves the problems with SIP peering and registering every 25 secs.
pcoventry
Grafter
Posts: 434
Registered: 26-11-2007

Re: VOIP Beta Trial Feedback

So far very impressive, a definate quality upgrade from broadband phone, well test other features over the weekend.
jakejohnson
Dabbler
Posts: 22
Registered: 06-11-2007

Re: VOIP Beta Trial Feedback

Quote from: ruslyn
The Draytek gives four options for the DTMF mode of the VOIP port - Inband, Outband (RFC2833), SIP INFO (cisco format) and SIP INFO (nortel format).  I was using the default option of 'Inbound' but by changing it to 'SIP INFO (cisco format)' the PIN is now being accepted and I have been able to record the voice prompt.
In case anybody else looking at this is also having the same problem with a Draytek 2800VG, or similar,  then you will need to change the DTMF mode by going to 'Phone Settings' under the VOIP menu. Then click on the port number that you are using followed by the 'Advanced' tab. The DTMF options are then shown.
That just leaves the problems with SIP peering and registering every 25 secs.
Hi ruslyn,
Our system uses DTMF RFC2833 so if you can try that one it may help.
As for your registration, you can try updating your setting and trying again. If this fails to work then you will need to contact Draytek support over this as it is an issue with the hardware.
In regards to SIP peering, if you can email/PM me (I'll PM my address) your extension and the number/s you are trying to dial, we can run some traces and see where the calls are failing.
Thanks,
Jake
acr
Rising Star
Posts: 217
Thanks: 11
Registered: 06-04-2007

Re: VOIP Beta Trial Feedback

I tried 'Outband (RFC2833)' and this did not work. SIP INFO (cisco format) was the first one on the list that worked.
I have sent you a PM with my extensions for Plusnet & Sipgate.
Thanks.
Moggy
Grafter
Posts: 99
Registered: 06-04-2007

Re: VOIP Beta Trial Feedback

Phew,
Not much chance for us non techie type chaps then!
The Draytek gives four options for the DTMF mode of the VOIP port - Inband, Outband (RFC2833), SIP INFO (cisco format) and SIP INFO (nortel format).  I was using the default option of 'Inbound' but by changing it to 'SIP INFO (cisco format)' the PIN is now being accepted and I have been able to record the voice prompt.
What!!!!!
acr
Rising Star
Posts: 217
Thanks: 11
Registered: 06-04-2007

Re: VOIP Beta Trial Feedback

Quote from: Jake

As for your registration, you can try updating your setting and trying again. If this fails to work then you will need to contact Draytek support over this as it is an issue with the hardware.

I have contacted Draytek support and supplied them with a copy of 'voip de se' from telnet to the router. They have responded as follows:-
"The SIP server (Gradwell) is telling your Vigor2800VG that the registration expires every 25 seconds.
We can see the "expires=25" in the message received back from the Gradwell. You can see from the voip de sh log that the Vigor2800VG is sending a message initially with an expiry of 600 seconds. If you change the settings in the Draytek then I'm sure you'll see the expiry change in the SIP message.
The Vigor2800VG is configured to obey the expiry that the SIP server is requesting. If they stop requesting an expiry of 25 seconds then it will extend the expiry."
Jake, your comment that this is a hardware fault would appear to be incorrect as it is the Gradwell SIP server which is providing the wrong expiry time.
Community Veteran
Posts: 6,262
Thanks: 433
Fixes: 40
Registered: 30-07-2007

Re: VOIP Beta Trial Feedback

If it helps , below is a log obtained from Sjphone.
It also shows 'expires=28'
From startup it seems to give 'expires=26' then next time 'expires=27' then settles down to 'expires=28'
Hope this is useful.
20:26:54 DEBUG   
2007-12-05 20:26:54.264 UDP 194.165.60.134:5082->LOCAL
SIP/2.0 200 OK
To: <sip:8408148@sip2.plus.net>;tag=238a7d19
From: <sip:8408148@sip2.plus.net>;tag=88092859321421
Via: SIP/2.0/UDP 192.168.5.64;branch=z9hG4bKc0a80540000000104757098e000008dc00000013
Call-ID: 944B32A8-97BD-4495-8C52-EAF903C490F1@192.168.5.64
CSeq: 9 REGISTER
Contact: <sip:8408148@80.229.236.203:5060>;expires=28
Content-Length: 0

20:26:54 INFO      Registration of sip:8408148@sip2.plus.net succeeded.
20:27:10 DEBUG   
2007-12-05 20:27:10.436 UDP LOCAL->194.165.60.134:5082
OPTIONS sip:nat.plus.net:5082 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.64;rport;branch=z9hG4bKc0a80540000000104757099e000064de00000015
Content-Length: 0
Call-ID: F8AA25F8-D350-4825-9438-ABC1DB723CEA@192.168.5.64
CSeq: 3 OPTIONS
From: <sip:8408148@sip2.plus.net>;tag=88094479630916
Max-Forwards: 70
To: <sip:nat.plus.net:5082>

20:27:22 INFO      Registration of sip:8408148@sip2.plus.net engaged...
20:27:22 DEBUG   
2007-12-05 20:27:22.264 UDP LOCAL->194.165.60.134:5082
REGISTER sip:sip2.plus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.5.64;rport;branch=z9hG4bKc0a8054000000010475709aa0000233c00000016
Content-Length: 0
Contact: <sip:8408148@80.229.236.203:5060>
Call-ID: 944B32A8-97BD-4495-8C52-EAF903C490F1@192.168.5.64
CSeq: 10 REGISTER
From: <sip:8408148@sip2.plus.net>;tag=88095662528302
Max-Forwards: 70
To: <sip:8408148@sip2.plus.net>
User-Agent: SJphone/1.60.289a (SJ Labs)

20:27:22 DEBUG   
2007-12-05 20:27:22.295 UDP 194.165.60.134:5082->LOCAL
SIP/2.0 200 OK
To: <sip:8408148@sip2.plus.net>;tag=20ec4a33
From: <sip:8408148@sip2.plus.net>;tag=88095662528302
Via: SIP/2.0/UDP 192.168.5.64;branch=z9hG4bKc0a8054000000010475709aa0000233c00000016
Call-ID: 944B32A8-97BD-4495-8C52-EAF903C490F1@192.168.5.64
CSeq: 10 REGISTER
Contact: <sip:8408148@80.229.236.203:5060>;expires=28
Content-Length: 0

20:27:22 INFO      Registration of sip:8408148@sip2.plus.net succeeded.
RichSmol
Grafter
Posts: 709
Registered: 29-10-2007

Re: VOIP Beta Trial Feedback

Can you please advise what issues you are getting which are caused by the short registration interval?