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SipDiscount END-OF-LIFE announcement for old IAX2/SIP server
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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
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"{{{{{Welcome}}}}}" <bhx___spam@trapped___hotmail.co.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.
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| paul123 2006-01-13, 8:47 pm |
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Jono wrote:
> 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.
Neither outgoing nor incoming on SIP? Or just incoming voip-in numbers?
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"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?
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| paul123 2006-01-13, 8:47 pm |
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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 with
> Asterisk?
That was my next question
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"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?
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| paul123 2006-01-13, 8:47 pm |
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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.....
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| paul123 2006-01-13, 8:47 pm |
| 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.
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| 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?
--
<http://ale.cx/> (AIM:troffasky) (gebssnfxl@ubgznvy.pbz)
15:35:56 up 13 days, 2:53, 2 users, load average: 0.79, 0.56, 0.57
This is my BOOOOOOOOOOOOOOOOOOOOOMSTICK
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| paul123 2006-01-14, 5:45 pm |
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alexd wrote:
> paul123 wrote:
>
>
> I'm not sure how they could stop you from making SIP-SIP calls, ie surely it
> just works?
>
I don't know. Isn't SIP-SIP something that would be have to be
"enabled" through their servers?
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| mark kelly 2006-01-14, 5:45 pm |
| No as soon as you use SIP1 the VOIP-in work, both birmingham and 056
numbers. shame Orange does not accept the 056 number.
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| paul123 2006-01-14, 5:45 pm |
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mark kelly wrote:
> No as soon as you use SIP1 the VOIP-in work, both birmingham and 056
> numbers. shame Orange does not accept the 056 number.
Yes the voip-in numbers work on sip1, but I meant SIP-SIP as in peering
to other networks like FWD, voiptalk etc,
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| Martin² 2006-01-14, 8:45 pm |
| Paul 123:
>but I meant SIP-SIP as in peering to other networks like FWD, voiptalk etc,
AFAIK they haven't got (any ?) peering agreements.
But you should be able to call anyone with SIP device by their IP number
(see my post few minutes earlier)
Regards,
Martin
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| paul123 2006-01-15, 7:45 am |
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Martin=B2 wrote:
> But you should be able to call anyone with SIP device by their IP number
Has anyone done this? And how about calling using their username - ie
"johnsmith" on xlite to "billnben" on an IP phone?
Paul
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| paul123 wrote:
>
> alexd wrote:
>
> I don't know. Isn't SIP-SIP something that would be have to be
> "enabled" through their servers?
No. Well sort of, they can disallow reinvites. All you have to do is tell
your SIP device to not use SipDiscount's SIP proxy. Then 'dial' the IP
address or hostname of the person you wish to call; presuming that they
accept calls from your IP address, then it'll Just Work. To be honest I
can't see why SipDiscount would stop you from making SIP-SIP calls through
their proxy, unless they want to make all calls chargeable, but that would
be overly cynical of me, wouldn't it? ;-)
As an aside: people seem to forget that SIP is a peer-to-peer protocol; the
'endpoints' [telephones in this case] know how to talk to each other, and
they don't require a central server ['exchange'] to do anything for them.
However, having the intelligence at the edge of the network isn't something
that telcos like [see previous paragraph], so they will try to foist
crippled solutions onto customers for as long as they can.
http://en.wikipedia.org/wiki/Sessio...iation_Protocol
--
<http://ale.cx/> (AIM:troffasky) (gebssnfxl@ubgznvy.pbz)
11:00:48 up 13 days, 22:18, 2 users, load average: 0.21, 0.30, 0.27
This is my BOOOOOOOOOOOOOOOOOOOOOMSTICK
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| paul123 wrote:
>
> Martin² wrote:
>
>
> Has anyone done this? And how about calling using their username - ie
> "johnsmith" on xlite to "billnben" on an IP phone?
Yes. I've had someone call my extension at my IP address from his softphone.
--
<http://ale.cx/> (AIM:troffasky) (gebssnfxl@ubgznvy.pbz)
18:30:56 up 14 days, 5:48, 2 users, load average: 0.38, 0.34, 0.30
This is my BOOOOOOOOOOOOOOOOOOOOOMSTICK
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| paul123 2006-01-15, 5:45 pm |
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alexd wrote:
> Yes. I've had someone call my extension at my IP address from his softphone.
Sorry to be a pain, not sure if I'm clear on your answer. Did they call
you by dialling 192*168*1*4 (or something similar) or did they
dial/type "alexd's username"?
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| paul123 wrote:
> alexd wrote:
>
>
> Sorry to be a pain, not sure if I'm clear on your answer. Did they call
> you by dialling 192*168*1*4 (or something similar) or did they
> dial/type "alexd's username"?
He was using a softphone, ie a SIP phone implemented as a software
application on his PC [http://www.linphone.org/?lang=us&rubrique=1 to be
precise]. He called my extension at my hostname, and it worked.
--
<http://ale.cx/> (AIM:troffasky) (gebssnfxl@ubgznvy.pbz)
19:03:55 up 23:18, 2 users, load average: 0.00, 0.01, 0.05
This is my BOOOOOOOOOOOOOOOOOOOOOMSTICK
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