[Spce-user] SIP trunk and TO header

[Sylvester Nielsen]

Hi,

I'm trying to get a subscriber to work as a SIP trunk for an Asterisk PBX.
The device contact address is sip:s at 87.116.27.XXX
I can register and send calls to the asterisk PBX, but when I call a number assigned to the subscriber the TO header address is sip:s at 87.116.27.XXX:5060 and not sip:did at 87.116.27.XXX which is needed to handle the routing correctly.

Is there a way to change this behavior per subscriber?


Med venlig hilsen

   Sylvester Nielsen

---------------------------------------------------------------

Fest Data A/S
Generatorvej 8D, 1.th
2730 Herlev

e-mail : sn at fest.dk<mailto:sn at fest.dk>

Tlf:     +45 4582 7389
Fax:   +45 4582 7387
Dir:    +45 6996 2011

http://www.festdata.dk

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