[Freeswitch-users] Conectin a SPA3102 as pstn gateway on FS

[Jaime Cruz]

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 at 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 at voip.hablaip.com>}user/
100123 at voip.hablaip.com
======
INVITE sip:100123 at 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 at voip.hablaip.com>;tag=XZvcHv1avcZ5c
To: <sip:226560600 at voip.hablaip.com>
Call-ID: 2b0f621f-b9c0-1233-ab8f-000c292ac960
CSeq: 79255480 INVITE
Contact: <sip:mod_sofia at 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 at voip.hablaip.com>]
==========



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

> Special assignment of used port is not required
> Using dial-string like "user/reg_user at mydomain.org" automatically do it
>
> Sergey
>
> On Sun, Aug 9, 2015 at 1:34 PM, <covici at ccs.covici.com> wrote:
>
>> Also, at least in mine, you have to use port 5061 to dial out.
>>
>>
>>
>
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-- 
Jaime Cruz
jaimecm at gmail.com
https://www.facebook.com/jaimecruzmendoza
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