[asterisk-users] additional range parameter for sip peer

2014-03-29 Thread Thomas Rechberger
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 11, biggest providers in Germany are

Re: [asterisk-users] Asterisk CLI Banner

2014-03-29 Thread Paul Belanger
On Fri, Mar 28, 2014 at 2:39 PM, Steve Edwards asterisk@sedwards.com 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

Re: [asterisk-users] Asterisk CLI Banner

2014-03-29 Thread Richard Kenner
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 pieces

[asterisk-users] handset forwarding Diversion header cannot be set on Local channels

2014-03-29 Thread Al lists
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 sip:202@192.168.1.46;reason=deflection Then asterisk sends the call to local channel: - Now forwarding SIP/201-0483 to