if you prefix the sofia dial string with sip: you should be able to pass anything you want.<br><br>sofia/internal/<span style="word-spacing: 0px; font-family: Helvetica; font-style: normal; font-variant: normal; font-weight: normal; font-size: 12px; line-height: normal; font-size-adjust: none; font-stretch: normal; text-transform: none; color: rgb(0, 0, 0); text-indent: 0px; white-space: normal; letter-spacing: normal; border-collapse: separate;"><a href="mailto:rn=9083820000@204.123.123.123:5060" target="_blank">sip:9085551212;npdi=yes;rn=9083820000@204.123.123.123:5060</a></span><br>
<br><div class="gmail_quote">2009/3/27 James H Thompson <span dir="ltr"><<a href="mailto:jht@lj.net">jht@lj.net</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div style="" bgcolor="#ffffff">
<div><font size="2">Calls would be sent to the IP address after the <a href="mailto:%27@%27" target="_blank">'@'</a> in the URI.</font></div>
<div><font size="2">Section 19.1.1 of RFC 3261 seems to say that TEL URIs can be
used as the user part of a SIP URI.</font></div>
<div><font size="2">My example Invite URI is the way we are receiving
traffic from some of the major telecom carriers.</font></div>
<div><font size="2">We would like be able to generate calls using the same
formats.</font></div>
<div> </div>
<div><font size="2"></font> </div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;">----- Original Message -----
<div style="background: rgb(228, 228, 228) none repeat scroll 0% 0%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial;"><b>From:</b> <a title="mike@jerris.com" href="mailto:mike@jerris.com" target="_blank">Michael Jerris</a> </div>
<div><b>To:</b> <a title="freeswitch-users@lists.freeswitch.org" href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>
</div>
<div><b>Sent:</b> Friday, March 27, 2009 5:41 AM</div>
<div><b>Subject:</b> Re: [Freeswitch-users] generating RFC 3966 and RFC 4694
calls</div></div>
<div><br></div>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?
<div><br></div>
<div>Mike</div>
<div><br>
<div>
<div>On Mar 27, 2009, at 6:11 AM, James H Thompson wrote:</div><br>
<blockquote type="cite"><span style="word-spacing: 0px; font-family: Helvetica; font-style: normal; font-variant: normal; font-weight: normal; font-size: 12px; line-height: normal; font-size-adjust: none; font-stretch: normal; text-transform: none; color: rgb(0, 0, 0); text-indent: 0px; white-space: normal; letter-spacing: normal; border-collapse: separate;">
<div bgcolor="#ffffff">
<div><font size="2">I need to generate calls with Invite URIs in this
format:</font></div>
<div> </div>
<div>INVITE<span> </span><a href="mailto:rn=9083820000@204.123.123.123:5060" target="_blank">sip:9085551212;npdi=yes;rn=9083820000@204.123.123.123:5060</a><span> </span>SIP/2.0</div>
<div><font size="2"></font> </div>
<div><font size="2">Is there an easy way to do this?</font></div>
<div><font size="2"></font> </div>
<div><font size="2">Thanks.</font></div>
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