SipDiscount END-OF-LIFE announcement for old IAX2/SIP server

Discussion in 'UK VOIP' started by Jono, Jan 13, 2006.

  1. Jono

    Jono Guest

    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
     
    Jono, Jan 13, 2006
    #1
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  2. Thus spaketh Jono:
    Received too the same roughly 3 hours ago.
     
    {{{{{Welcome}}}}}, Jan 13, 2006
    #2
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  3. Jono

    Jono Guest

    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.
     
    Jono, Jan 13, 2006
    #3
  4. Jono

    paul123 Guest

    Neither outgoing nor incoming on SIP? Or just incoming voip-in numbers?
     
    paul123, Jan 13, 2006
    #4
  5. Jono

    Jono Guest

    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?
     
    Jono, Jan 13, 2006
    #5
  6. Jono

    paul123 Guest

    I have [email protected] 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 was my next question
     
    paul123, Jan 13, 2006
    #6
  7. Jono

    Jono Guest

    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.
    ...but what's the answer?
     
    Jono, Jan 13, 2006
    #7
  8. Jono

    paul123 Guest

    Too right, a long call to my mobile is pencilled in my diary for Jan
    30!
    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:p/username
     
    paul123, Jan 13, 2006
    #8
  9. Jono

    paul123 Guest

    getting back to your original post, and I've had one of these emails
    since too, they say:
    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.
     
    paul123, Jan 14, 2006
    #9
  10. Jono

    alexd Guest

    I'm not sure how they could stop you from making SIP-SIP calls, ie surely it
    just works?
     
    alexd, Jan 14, 2006
    #10
  11. Jono

    paul123 Guest

    I don't know. Isn't SIP-SIP something that would be have to be
    "enabled" through their servers?
     
    paul123, Jan 14, 2006
    #11
  12. Jono

    mark kelly Guest

    No as soon as you use SIP1 the VOIP-in work, both birmingham and 056
    numbers. shame Orange does not accept the 056 number.
     
    mark kelly, Jan 14, 2006
    #12
  13. Jono

    paul123 Guest

    Yes the voip-in numbers work on sip1, but I meant SIP-SIP as in peering
    to other networks like FWD, voiptalk etc,
     
    paul123, Jan 14, 2006
    #13
  14. Jono

    Martin² Guest

    Paul 123:
    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
     
    Martin², Jan 15, 2006
    #14
  15. Jono

    paul123 Guest

    Has anyone done this? And how about calling using their username - ie
    "johnsmith" on xlite to "billnben" on an IP phone?

    Paul
     
    paul123, Jan 15, 2006
    #15
  16. Jono

    alexd Guest

    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/Session_Initiation_Protocol
     
    alexd, Jan 15, 2006
    #16
  17. Jono

    alexd Guest

    Yes. I've had someone call my extension at my IP address from his softphone.
     
    alexd, Jan 15, 2006
    #17
  18. Jono

    paul123 Guest

    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"?
     
    paul123, Jan 15, 2006
    #18
  19. Jono

    alexd Guest

    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.
     
    alexd, Jan 16, 2006
    #19
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