On Mar 28, 2009, at 4:05 AM, Anthony Minessale wrote:
if you prefix the sofia dial string with sip: you should be able to pass anything you want.
sofia/internal/sip:9085551212;npdi=yes;rn=9083820000@204.123.123.123:5060
Is that similar as this? <action application="bridge" data="sofia/sip/9998881111@sip.yourprovider.com "/> got it from wiki: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#From_the_Dia...
2009/3/27 James H Thompson <jht@lj.net> Calls would be sent to the IP address after the '@' in the URI. Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be used as the user part of a SIP URI. My example Invite URI is the way we are receiving traffic from some of the major telecom carriers. We would like be able to generate calls using the same formats.
----- Original Message ----- From: Michael Jerris To: freeswitch-users@lists.freeswitch.org Sent: Friday, March 27, 2009 5:41 AM Subject: Re: [Freeswitch-users] generating RFC 3966 and RFC 4694 calls
You seem to be confusing your standards, those 2 specs are about tel: uri's not sip: uris. Sending a tel uri I am not sure we can do, where would we send it to?
Mike
On Mar 27, 2009, at 6:11 AM, James H Thompson wrote:
I need to generate calls with Invite URIs in this format:
INVITE sip:9085551212;npdi=yes;rn=9083820000@204.123.123.123:5060 SIP/2.0
Is there an easy way to do this?
Thanks.
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