Hi Sergey,

Thanks for your help,
it' worked to change the Invite "To:" Uri,

but as I tested, the linksys SPA3000&3102 is not taken the destination from the To Uri nor from the Invite Uri as "INVITE sip:nnnnnn@200.200.200.200:5061 SIP/2.0" as "nnnnnn" can be any destination that I want to dial via it's PSTN line.

Do you think that there's an option to change this in FS,

Thanks for your help.

JCM

======

originate: {sip_invite_to_uri=<sip:226560600@voip.hablaip.com>}user/100123@voip.hablaip.com

======
INVITE sip:100123@200.200.200.200:5061 SIP/2.0
Via: SIP/2.0/UDP 100.100.100.100;rport;branch=z9hG4bK32ScZ5j0X2NQH
Max-Forwards: 69
From: "Katty Casa." <sip:Katty Casa@voip.hablaip.com>;tag=XZvcHv1avcZ5c
To: <sip:226560600@voip.hablaip.com>
Call-ID: 2b0f621f-b9c0-1233-ab8f-000c292ac960
CSeq: 79255480 INVITE
Contact: <sip:mod_sofia@100.100.100.100:5060>
======
2015-08-10 02:04:49.318695 [DEBUG] switch_ivr_originate.c:2101 Parsing global variables
2015-08-10 02:04:49.318695 [DEBUG] switch_event.c:1698 Parsing variable [sip_invite_to_uri]=[<sip:226560600@voip.hablaip.com>]
==========

On Sun, Aug 9, 2015 at 11:43 AM, Sergey Safarov <s.safarov@gmail.com> wrote:

Special assignment of used port is not required
Using dial-string like "user/reg_user@mydomain.org" automatically do it

Sergey

On Sun, Aug 9, 2015 at 1:34 PM, <covici@ccs.covici.com> wrote:
Also, at least in mine, you have to use port 5061 to dial out.

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Jaime Cruz
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