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FYI Free SIP URI dialing service for your own domain

Discussion in 'Add-On Install Instructions' started by w1ve, Aug 23, 2013.

  1. w1ve Guru

    lgaetz and wardmundy like this.
  2. wardmundy Nerd Uno

    Last edited by wardmundy, Aug 24, 2013
  3. lgaetz Pundit

    I don't know much about DNS as will be evident by these questions:
    Is there a possibility of conflict if I use the same URI as email address; i.e. call or email me at lgaetz@example.com?

    The other thing is my domain registrar, Netfirms only gives me one field to enter a TXT record, I can't enter a name/host. I will open a ticket, but wondered if someone out there knows how to use this.
  4. wardmundy Nerd Uno

    Entries are not actually the same. Even though both email and SIP are addressed as ward AT mundy.org, in DNS, the entries would be different. One would be ward while SIP is sip-ward. Email doesn't take its user entries from DNS anyway. They're actually stored in the mail server. Better analogy would be a web site, e.g. ward.mundy.org and SIP entry of ward AT mundy.org can still coexist. I think that's how it works anyway. :smartass:
    Last edited by wardmundy, Aug 24, 2013
  5. w1ve Guru

    Exactly, Ward. A Cool implementation... when you make a SIP call, the SRV record points to the sipcloak.org server. When he gets the SIP INVITE, he does a DNS lookup for a TXT record matching 'sip-YourAccountBefore@Sign' and sends the call to the address you specify. A very cool idea!
  6. w1ve Guru

    BTW, cloudns.net is one of the free DNS services I use. They have lots of geographic diversity, and paid accounts are very cheap. They also support all the record types with an easy-to-use web interface. My absolute fav is dyn.com, but they cost $$$ -- I have a bunch of free accounts for life that were ported when they bought another company I was using. :)
    wintek and wardmundy like this.
  7. wardmundy Nerd Uno

    In addition to lenny, for any Gurus that would like a "vanity SIP URI" @pbxinaflash.com, just PM me your sip2sip.info number. Once we get the kinks ironed out, I don't mind opening it up for everyone else. Among other things, it gives us an easy way to contact each other at no cost.
    Last edited by wardmundy, Aug 24, 2013
  8. lgaetz Pundit

  9. wardmundy Nerd Uno

    Darnit. Somebody already got that one but, if you'd like the calls, we can work something out. :death:
  10. wardmundy Nerd Uno

    Thanks to the genius of Bill Simon, the magic of YATE, and the generosity of RentPBX, we now have our own DNS cloaking servers running on the East and West Coast for those that prefer routing your SIP calls through a known quantity. You can either use your own domain or, for PIAF Gurus, send me your SIP URI and we'll add a Vanity Entry for yourname@pbxinaflash.com.

    If you're hosting your own DNS server, here are the entries to add redundant cloaking assuming the desired SIP URI for calls to 123@sip2sip.info is bill@yourdomain.com:
    _sip._udp.yourdomain.com. IN SRV 10 10 5060 east.pbxinaflash.com.
    _sip._udp.yourdomain.com. IN SRV 10 10 5060 west.pbxinaflash.com.
    sip-bill.yourdomain.com.  IN TXT "123@sip2sip.info"
    For those with a shared hosting account that uses cPanel and WHM (such as one of our sponsors, BlueHost), this takes about 2 seconds to implement.

    Nerd Vittles tutorial on creating a free, secure SIP URI using sip2sip.info: http://nerdvittles.com/?p=6914
    Last edited by wardmundy, Aug 25, 2013
    w1ve and lgaetz like this.
  11. wardmundy Nerd Uno

    [IMG] Why do you want a secure SIP URI for your Asterisk server? Because all calls worldwide to SIP URIs using Asterisk, YATE, FreeSwitch or any SIP phone are free!
    Last edited by wardmundy, Aug 25, 2013
  12. w1ve Guru

  13. wardmundy Nerd Uno

  14. w1ve Guru

    ok... I'm not nuts... Will ask them why.
  15. VaHam Member

    I think the success or failure lies in how the 302 "moved temporarily" is handled by the client. For instance I found the cloaking thru both sipcloak and pbxinaflash failed using blink and the kitchen sink also. However Jitsi shows the 302 but proceeds to connect the call.
    wardmundy likes this.
  16. VaHam Member

    Ok this is interesting. I did some more testing by setting up an onsip sip account and while using that blink connects the call when using pbxinaflash cloak. I'd have to examine the difference in packet responses to see why jitsi works with both sip2sip and onsip blink only works when using onsip (when using cloaked links) but there must be differences in how the 302 moved call transfer is being presented to the softphone.

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