[Asterisk-Users] Dynamic SIP outbound usernames?

[Zac Sprackett]

Add dial plan entries like this in extensions.conf

[trunks-ld]
; long distance
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@mt-1204,20,Tr)
exten => _1NXXNXXXXXX,2,Congestion

add a sip entry like this in sip.conf

[mt-1204]
type=peer
host=172.20.16.7
mask=255.255.255.255
dtmfmode=inband
context=default
qualify=yes
canreinvite=no

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Alastair Maw
> Sent: Tuesday, September 09, 2003 5:27 AM
> To: * Users List
> Subject: [Asterisk-Users] Dynamic SIP outbound usernames?
> 
> 
> Hi,
> 
> I have * set up as a PSTN->VoIP gateway (with an E1 with multiple 
> numbers pointing to it).
> 
> I'd really like to be able to dial out to a SIP server like so:
> 
>    exten => _X.,1,Dial(SIP/${DNID}@hostname)
> 
> I.e. the remote SIP server receives a SIP INVITE with a "To:" header 
> containing the dialed number (e.g. 02085555555 at computer.company.com).
> 
> This is equivalent to having a hostname extension in sip.conf with a 
> dynamic username of ${DNID}.
> 
> How does one achieve this?
> 
> Likewise, it would be nice to be able to use gnophone to simulate calls 
> into the system, by pointing it at the * box and getting the dialed 
> number on that to route things in the same way.
> 
> Any ideas?
> 
> -- 
> Alastair Maw <al.maw at mxtelecom.com>
> MX Telecom - Systems Analyst
> http://www.mxtelecom.com
> 
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