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Unread 08-09-2007, 07:06 PM   #1
keschrich
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Default Standard format for representing SIP numbers?

I'm just getting started with using pure SIP (I've been using Asterlink at work for quite some time to terminate calls from my * system)..

Anyway, I now have a SIP number on FWD, and would like to put that on my business card; is there a standardized format to use that would be easily understood by most?

i.e:

SIP: 805xxx (FWD) or maybe
SIP: *393-805xxx (SIPBroker)

or do enough SIP services use e164.org that I could just list my regular number and tell people that it will carry over the internet if they're using voip as well?

How do the rest of you advertise your numbers?

Thanks for putting up with such a boneheaded question
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Unread 08-09-2007, 10:05 PM   #2
kurun
 
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Interesting question. I have listed mine simply as SIP: *010-xxxxxx

I figure if anybody knows anything about SIP peering, and SIP-Broker access numbers, they will know what to do with that.

If they don't, well they will have to use my regular PSTN number .....
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Unread 08-09-2007, 10:09 PM   #3
kieranmullen
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I have not seen many use either one! I use enum from e614.org and siprbroker when possible.

Why not put your sip URI down on your card or better yet

yourregulartelephone@yourdomain.com
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Unread 08-11-2007, 08:38 AM   #4
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Quote:
Originally Posted by kieranmullen View Post
I have not seen many use either one! I use enum from e614.org and siprbroker when possible.

Why not put your sip URI down on your card or better yet

yourregulartelephone@yourdomain.com
I am sure you will find a good response .
Good luck for you.
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Unread 08-16-2007, 07:25 AM   #5
mberlant
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My card lists my various SIP and two-stage DID services two ways. I have

FWD 4xxxxx
SIPBroker *7xxx-xxxx#
BBPGlobal 111xxxx

and also

+1 416 548-8458 Ext. *7xxx-xxxx# (Toronto)
+61 3 8687-5478 Ext. 111xxxx (Melbourne)

When closing an email to someone, I always give the person the number closest to him or her to facilitate a return call. Vendors and customers are much quicker to pick up the phone when it's a "local" call.

I don't include SIP:number@domain because it's not at all easy to place such a call from most SIP phones and ATAs.
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