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[jht at lj.net]

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.

[brian at freeswitch.org]

Params are not added before the @ sign that I'm pretty sure of.

<action application="export" data="sip_invite_params=this=kewl;that=iskewl"/>
<action application="bridge" data="sofia/profile/blah@blah"/>




Results in INVITE [url=sip:blah@blah;this=kewl;that=iskewl]sip:blah@blah;this=kewl;that=iskewl[/url] SIP/2.0




You also have sip_invite_to_params, sip_invite_from_params, sip_invite_contact_params


/b





On Mar 27, 2009, at 5:11 AM, James H Thompson wrote:

[mike at jerris.com]

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:

[jht at lj.net]

Calls would be sent to the IP address after the '@' ([email]'@'[/email]) 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 (mike@jerris.com) To: freeswitch-users@lists.freeswitch.org (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 ([email]rn=9083820000@204.123.123.123:5060[/email]) SIP/2.0 Is there an easy way to do this? Thanks. _______________________________________________Freeswitch-users mailing listFreeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _______________________________________________Freeswitch-users mailing listFreeswitch-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org

[kristian.kielhofner at...]

Some platforms add/use URI params in the user portion of the URI.

I was just reminded of this this week with Global Crossing:

sip:+18005551212;npdi=yes@0.0.0.0

That is (of course) the number portability display indicator...

Anyways, I had to rewrite the URI using OpenSIPS to strip the
";npdi=yes". I imagine you could do the same but I'm not sure how in
FreeSWITCH...

2009/3/27 Brian West <brian@freeswitch.org>:

[brian at freeswitch.org]

Thats about dumb... um I don't think we let you put things in there.. but it would only be a few lines to do it...blah rfc's suck!

/b

On Mar 27, 2009, at 2:47 PM, Kristian Kielhofner wrote:

[anthony.minessale at g...]

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 ([email]rn=9083820000@204.123.123.123:5060[/email])

2009/3/27 James H Thompson <jht@lj.net (jht@lj.net)>

[dujinfang at gmail.com]

On Mar 28, 2009, at 4:05 AM, Anthony Minessale wrote:

[brian at freeswitch.org]

no

sofia/profile/[url=sip:blah@blah]sip:blah@blah[/url]


sip: makes sofia take it as is.


/b



On Mar 27, 2009, at 9:39 PM, dujinfang wrote: