is there anyway to change Sip headers in local channels?
if a user sets forward on their handset, calls coming in to the handset get
diversion header added:
Diversion: "202" ;reason=deflection
Then asterisk sends the call to local channel:
- Now forwarding SIP/201-0483 to 'Local/33@tes
> If you really want to do it:
>
> 1) create a wrapper to asterisk -r
> 2) pipe the welcome message to /dev/null
> 3) ???
> 4) profit
>
> you didn't modify Asterisk.
No you didn't, but you may neverthess have created a derived work. There
are two different legal arguments you can make when two
On Fri, Mar 28, 2014 at 2:39 PM, Steve Edwards
wrote:
> On Fri, 28 Mar 2014, Richard Kenner wrote:
>
>> And this certainly may vary from jurisdiction to jurisdiction. For a
>> (quite dated at this point) discussion of this issue from a US perspective,
>> see
>>
>>
>> http://www.law.berkeley.edu/p
Many ITSP are using loadbalancers, so if somebody registers on a sip
peer with specific dns host, an incoming call may be received from a
different ip and the host value in peer section doesnt match, so it will
go to default context.
For example Telekom or 1&1, biggest providers in Germany are
