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 Posted: Tue Feb 17, 2015 6:10 pm Post subject: [asterisk-users] How to route SIP provider without DID |
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Am 17.02.2015 um 23:44 schrieb hbk:
| Quote: | 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
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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@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
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