[asterisk-users] How to route SIP provider without DID

[Markus]

Am 17.02.2015 um 23:44 schrieb hbk:
> I'm struggling to separate inbound calls fro a SIP provider that does
> not send DID.
> I have tried  .......sip.com/12345678 on register string
> different context=from-no-did
> Port not possible as only support 5060

You're right, this is always an annoying and confusing scenario. Here's 
my sample for sipgate which works for separating inbound and outbound:

sip.conf:

register => user:pass at sipgate.de/sipgate-in

[sipgate-out]
type=friend
insecure=invite
nat=no
username=user
fromuser=user
fromdomain=sipgate.de
secret=pass
host=sipgate.de
qualify=no
canreinvite=no
dtmfmode=rfc2833
context=sipgate

extensions.conf:

[sipgate]
exten => sipgate-in,1,NoOp
exten => sipgate-in,n,Dial(SIP/priv)

(This is for incoming calls only)

And for my SIP hardphone which receives the calls from sipgate and dials 
out via sipgate:

sip.conf:

[priv]
type=friend
secret=anotherpass
host=dynamic
nat=no
disallow=all
allow=ulaw
allow=alaw
canreinvite=no
context=sipgate-priv

[sipgate-priv]
exten => _X.,1,NoOp
exten => _X.,n,Dial(SIP/${EXTEN}@sipgate-out)


Good luck,
Markus