I'd ask provider to make a change and send destination number in SIP URI like most others do. Especially if later you get different numbers from a different provider, you might benefit from having it all standard from day 1. On 16-09-07 11:38 AM, Allan Kristensen wrote:
Hello,
I've just setup a sip trunk with a new provider, however there are some issues receiving calls.
The INVITE looks like this:
INVITE sip:NCdWXsdSsaF2B@192.168.0.190:52284 <http://sip:NCdWXsdSsaF2B@192.168.0.190:52284> SIP/2.0 .... From: <sip:+XXXXXXXXXX@x.x.x.x>;tag=Ud8ghaXg01821203 To: <sip:+YYYYYYYYYY@x.x.x.x> ...
The invite has my account name, instead of the destination number and the real destination number is in the "To:" field (which it normally is I guess ;-)
Then Freeswitch does this of cause:
[INFO] mod_dialplan_xml.c:637 Processing +XXXXXXXXXX <+XXXXXXXXXX>->NCdWXsdSsaF2B in context inbound
That is undesirable, so Is there any way for Freeswitch to grab the destination number from the to field instead of the INVITE string ? I could make a special sip profile for that provider with it's own dialplan and then set destination_number from the "sip_to_user" variable and transfer to the real dialplan, but it just seems so messy... Or is it my provider that is doing it wrong?
Best regards, Allan
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