This is propably because you've registered in sip.conf without an
extension in the register= configuration
line. Check the syntax.
If you can't fix that, use the SIP_HEADER() dialplan funcion to get
the value of the To: header and
use that as an extension.
/Olle
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* Olle E Johansson -
We are using 2 different incoming trunks.
The first one is alsion.com and is sending INVITE with phone number in
the INVITE line whereas plugandtel put the callee number only inside the
To: Section.
Marco Mouta a écrit :
Could you describe in detail how did you fall into this situation, I mean
Hi,
I would suggest you to use Asterisk Application SIPGetHeader in your
Dialplan for incoming calls from plugandtel.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPGetHeader
Something like
*
exten=_[a-z].,1,SIPGetHeader(Var_TO=To)
exten=_[a-z].,2,Dial(SIP/${Var_TO})
Please be
Good evening!
I was wondering one thing,
I'm using freepbx to configure my asterisk server and I have a problem
with some inbound calls.
When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an
inbound route! It matches a DID number.
How can I route an INVITE sip:[EMAIL PROTECTED]
Could you describe in detail how did you fall into this situation, I mean
the real example which SIP phone sends this invite? Is registered in
asterisk? it is a non-registered sip phone trying to dial a sip user at your
* box?
If this is an issue with a specific hardware outside of your asterisk,
