Just got the email below.

**************************

Hello!

We see you are still making calls to our old SIP server. Therefore you
receive this message

Soon we will shut down our old asterisk servers. The exact shutdown date has
not been decided yet but will be some time soon.
We are sending this email now so you have plenty of time to move to our new
SIP system.
When a shutdown-date has been decided then we will announce the date again.

How to move:

Connect your SIP device or softphone to our new SIP server with the same
username & password as you are using now

the old sip server you are connecting to now: sip.sipdiscount.com
the new sip server you have to change to:     sip1.sipdiscount.com

On the new SIPserver there is no GSM codec support for now, but this WILL be
supported before we shutdown the old servers, in the meantime use G711alaw
(no ulaw!) or G726

the advantage of the new SIP server:

- VOIP-IN SUPPORT!!
- better audio quality!
- soon: codec support for iLBC / G729 / G723 / GSM
- soon: P2P calls


For IAX users:

We know some of you asterisk-users will not be happy with it but IAX will
reach end-of-life soon too.
So if you use asterisk and use IAX to make calls, migrate to SIP!

Regards,
SipDiscount

[ Post a follow-up to this message ]

Thus spaketh Jono:
>
> SipDiscount END-OF-LIFE announcement for old IAX2/SIP server
>

Received too the same roughly 3 hours ago.


--
For £5 when referred to easyMobile contact me via
http://www.south-east-birmingham.tk
www.dodgy-dealer.co.uk

[ Post a follow-up to this message ]

"{{{{{Welcome}}}}}" <bhx___spam@trapped___hotmail.c
o.uk> wrote in message
news:A8Rxf.276$wl.197@text.news.blueyonder.co.uk...
> Thus spaketh Jono: 
>
> Received too the same roughly 3 hours ago.
>
>
> --
> For £5 when referred to easyMobile contact me via
> http://www.south-east-birmingham.tk
> www.dodgy-dealer.co.uk

Interestingly, I'm using the Voipbuster IAX servers, not Sipdiscount SIP/IAX
servers. I can't, for the life of me, get sip working at all on their
servers.

[ Post a follow-up to this message ]

Jono wrote:

> Interestingly, I'm using the Voipbuster IAX servers, not Sipdiscount SIP/I
AX
> servers. I can't, for the life of me, get sip working at all on their
> servers.

Neither outgoing nor incoming on SIP? Or just incoming voip-in numbers?

[ Post a follow-up to this message ]

"paul123" <paul@redy.net> wrote in message
news:1137179469.727830.317830@z14g2000cwz.googlegroups.com...
>
> Jono wrote:
> 
>
> Neither outgoing nor incoming on SIP? Or just incoming voip-in numbers?
>

Yeah, I should've expanded a little.

I settled on using the voipbuster IAX servers as SIP on Sipdiscount's
servers gave me one way audio problems.

I'm using voipstunt now, anyway.

That's an interesting point, how does one configure the voip-in numbers with
Asterisk?

[ Post a follow-up to this message ]

Jono wrote:
> "paul123" <paul@redy.net> wrote in message
> news:1137179469.727830.317830@z14g2000cwz.googlegroups.com... 
>
> Yeah, I should've expanded a little.
>
> I settled on using the voipbuster IAX servers as SIP on Sipdiscount's
> servers gave me one way audio problems.
>
> I'm using voipstunt now, anyway.
>

I have asterisk@home working with sipdiscount configured on a SIP trunk
to the sip1.sipdiscount server with no problems... (based on your post
a couple of weeks back)

> That's an interesting point, how does one configure the voip-in numbers wi
th
> Asterisk?

That was my next question

[ Post a follow-up to this message ]

"paul123" <paul@redy.net> wrote in message 
>
> I have asterisk@home working with sipdiscount configured on a SIP trunk
> to the sip1.sipdiscount server with no problems... (based on your post
> a couple of weeks back)

Doesn't like it here - I'm definitely using iax.voipbuster.com..at least
until they pull the plug. Must make sure I use up the credit.
 
>
> That was my next question
>

..but what's the answer?

[ Post a follow-up to this message ]

Jono wrote:

> Doesn't like it here - I'm definitely using iax.voipbuster.com..at least
> until they pull the plug. Must make sure I use up the credit.

Too right, a long call to my mobile is pencilled in my diary for Jan
30!

> ..but what's the answer?

give up.

If I had an incoming number to experiment with, I'd try the
following....
-----------
Incoming:
User context: Username
User details:
context=from-trunk
dtmfmode=info
insecure=very
secret=password
type=peer

Reg string:
username:password@sip1.sipdiscount.com/username
------------
and add an inbound route with the username
------------

No idea if it'll work, it's an adaptation of my sipgate settings and
will probably need tweaking.....

[ Post a follow-up to this message ]

getting back to your original post, and I've had one of these emails
since too, they say:
> the advantage of the new SIP server:
>
> - VOIP-IN SUPPORT!!
> - better audio quality!
> - soon: codec support for iLBC / G729 / G723 / GSM
> - soon: P2P calls

so, we'll have to wait for the change to be able to get voip-in to
work?

and does that "- soon: P2P calls" mean SIP-SIP? - that'd be 'andy
'arry.

[ Post a follow-up to this message ]

paul123 wrote:

> and does that "- soon: P2P calls" mean SIP-SIP? - that'd be 'andy
> 'arry.

I'm not sure how they could stop you from making SIP-SIP calls, ie surely it
just works?

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[ Post a follow-up to this message ]