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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