[asterisk-users] additional range parameter for sip peer
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 using too many different addresses that its not practical to define them all (up to 50 hosts and they still add!), as this will also generate too much traffic (especially with qualify and multiple registrations) and they may even lock you out as untrusted, which may even result in that they will block asterisk permanently for everybody. Thats not really desirable. I think its also not recommended in terms of security to use default context with allowguest=yes and sort the incoming calls by header, because this can be faked easily. From my understanding the permit/deny parameters are only used for incoming calls if host is set to dynamic and then there will be no outgoing registration to remote peer possible. permit/deny is used for access, not for matching. How about an additional parameter where an range of ip addresses can be defined in peer section, which will be used for matching calls? hostmatchrange=x.x.x.x/24 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI Banner
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 http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157 The publication (43 pages) is dated 1988. The DMCA (1998) and subsequent legislation may have changed the landscape. My (ignorant) opinion -- just don't. Is it worth the effort to research? Is it worth paying a lawyer to research it and give an opinion that may be worth nothing until it is examined in court? If you want to display something custom, how about a 'wrapper' script that displays a file using 'curl' before handing off to Asterisk -- easier to implement, easier to maintain, no legal BS to consider. Or can you express your creativity by fiddling with ASTERISK_PROMPT? 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. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI Banner
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 of software are tightly coupled in that way: one argues that it's a derived work and the other that it's not. Copyright law when it comes to software is not simple and certainly not obvious. If you want to use a piece of Free Software in a commercial product, you need to consult an attorney. It's really that simple. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] handset forwarding Diversion header cannot be set on Local channels
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 'Local/33@test' (thanks to SIP/202-0484) and not all Telco providers handle diversion header gracefully, some dont like to see 202 in header. i tried to set the sip header in target 33@test but asterisk see's this as local channel and wont do sip add header: WARNING[13584]: chan_sip.c:20562 func_header_read: This function can only be used on SIP channels. is there anyway around this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
