Re: [Asterisk-Users] sip URL peering

2005-11-23 Thread Klaus Darilion
There is a new ietf WG to come which deals with peering issues. It's 
called SPEER (formerly VOIPEER)


The list archive is at
http://darkwing.uoregon.edu/~llynch/voipeer/

minutes from last ietf meeting:
http://www3.ietf.org/proceedings/05nov/minutes/voipeer.html

regards
klaus

Chris Hills wrote:

Wolfgang S. Rupprecht wrote:


One thing I haven't seen get much airtime on the digium lists is sip
URL-based peering.  I imagine many of us have far more asterisk
extensions than PSTN numbers.  It would be really nice to be able to
do something like call [EMAIL PROTECTED] from [EMAIL PROTECTED]  It
looks like all or most of the pieces are in place, but I don't see
folks discussing it much.  Is no-one else interested in this?

  


Perhaps you would be interested in TRIP (telephony routing over ip)? 
Each organisation can apply for an ITAD number, just like a domain. TRIP 
numbers take the form *, for example, 1234*222. As you 
can no doubt surmise, TRIP numbers can be dialled from a regular 
telephone handset. For more information, please see the following 
documents:-


http://www.iana.org/assignments/trip-parameters
http://www.ietf.org/rfc/rfc3219.txt

Regards

--
Chris Hills
IT Services
North East Worcestershire College




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Re: [Asterisk-Users] sip URL peering

2005-11-22 Thread Chris Hills

Wolfgang S. Rupprecht wrote:

One thing I haven't seen get much airtime on the digium lists is sip
URL-based peering.  I imagine many of us have far more asterisk
extensions than PSTN numbers.  It would be really nice to be able to
do something like call [EMAIL PROTECTED] from [EMAIL PROTECTED]  It
looks like all or most of the pieces are in place, but I don't see
folks discussing it much.  Is no-one else interested in this?

  
Perhaps you would be interested in TRIP (telephony routing over ip)? 
Each organisation can apply for an ITAD number, just like a domain. TRIP 
numbers take the form *, for example, 1234*222. As you 
can no doubt surmise, TRIP numbers can be dialled from a regular 
telephone handset. For more information, please see the following 
documents:-


http://www.iana.org/assignments/trip-parameters
http://www.ietf.org/rfc/rfc3219.txt

Regards

--
Chris Hills
IT Services
North East Worcestershire College




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[Asterisk-Users] sip URL peering

2005-11-22 Thread Wolfgang S. Rupprecht

One thing I haven't seen get much airtime on the digium lists is sip
URL-based peering.  I imagine many of us have far more asterisk
extensions than PSTN numbers.  It would be really nice to be able to
do something like call [EMAIL PROTECTED] from [EMAIL PROTECTED]  It
looks like all or most of the pieces are in place, but I don't see
folks discussing it much.  Is no-one else interested in this?

One group that seems to have an ever growing list of sip servers that
accept direct incoming sip calls is sipbroker.  Using their service
doesn't really buy the average asterisk admin much, but they do have a
nice list of sip servers and they do assign a unique prefix code to
each server which might be useful to snarf into an asterisk database.

   http://www.sipbroker.com/sipbroker/action/providerWhitePages

extensions.conf:

;; send everything else with a ** prefix to Sipbroker 
;; strip one of the stars since they only want one in total. 
;;   http://www.sipbroker.com/sipbroker/action/providerWhitePages
exten => _**XXX.,1,Macro(dial-outgoing,SIP/${EXTEN:[EMAIL PROTECTED])

sip.conf:

;;; outgoing to Sipbroker
[sipbroker-out]
type=peer
host=sipbroker.com

Is sipbroker just a well-kept secret from the asterisk crowd, or is
everyone else using asterisk for phone spamming from call centers and
the last thing they want is folks to be able to call them back and
give them an earful over disturbing their dinner?

-wolfgang
-- 
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
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