[Freeswitch-users] generating RFC 3966 and RFC 4694 calls

[dujinfang]

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 at 204.123.123.123:5060
>

Is that similar as this?

<action application="bridge" data="sofia/sip/9998881111 at sip.yourprovider.com 
"/>

got it from wiki:  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#From_the_Dialplan
> 2009/3/27 James H Thompson <jht at 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 at 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 at 204.123.123.123:5060  
>> SIP/2.0
>>
>> Is there an easy way to do this?
>>
>> Thanks.
>>
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>
>
>
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> -- 
> Anthony Minessale II
>
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