similar to: TLS, SRTP, Asterisk11 and Snom870s

Am 03.03.2015 um 18:16 schrieb James B. Byrne: > CentOS-6.5 (FreePBX-2.6) > Asterisk-11.14.2 (FreePBX) > snom870-SIP 8.7.3.25.5 > > I am having a very difficult time attempting to get TLS and SRTP > working with Asterisk and anything else. At the moment I am trying to > get TLS functioning with our Snom870 desk-sets. And I am not having > much luck. > > Since this

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten

On Tue, March 3, 2015 13:37, James Cloos wrote: >>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: > > JBB> tcpenable=yes > JBB> tlsenable=yes > JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt > JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt > JBB> tlsdontverifyserver=yes > JBB>

>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: JBB> tcpenable=yes JBB> tlsenable=yes JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB> tlsdontverifyserver=yes JBB> tlscipher=ALL JBB> tlsclientmethod=tlsv1 You are missing the tls key. The config name is

Other things to consider: The transport config, which can be in [general] or in a peer's [] block. if you want tls-only, use transport=tls it also accepts tcp, udp or a comma-separated list. if given a list, it tries them in order If you need ast to register over tls, use something like this: register => tls://username:xxxxxx at sip-tls-proxy.example.org (copied from the

Arch x86_64 OS CentOS-6.4 (freepbx) Asterisk 11.4 FreePBX 2.11.0.4 Snom870 with FW-8.7.4.8 What I am attempting to do is to set a different background colour for the BLF vkeys when a station is set to DND. This is supposedly accomplished through this setting in the phones provisioning file: <vkey_blue perm="RW"> DND Blue.on Blue.pickup Blue.park Blue.message </vkey_blue>

I have thsi code in a dial plan. The purpose of which is to set distinctive ring tones for internal and transferred calls. exten => _.,1,Noop(CALLERID_ALL=${CALLERID(all)}) exten => _.,n,Set(CallerIDNum=${CALLERID(num)}) ; This just shows a list of interesting variables and their values ; Comment it out when finished debugging ;include => macro-dumpvars ;exten =>

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 We use Snom870 handsets with firmware v.8.7.3.19. I am trying to develop a custom dial plan to invoke a distinctive ring-tone when an external call is transferred internally. Based on an earlier solution I discovered I am attempting this: [from-internal] include => set-alert-if-local [from-internal-original]

can i test my asterisk11 on a single machine on which asterisk is installed that sounds are working from both end properly. i have installed asterisk 11 on ubuntu12.04 with twinkle soft phone. regards abhi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130216/40fa976d/attachment.htm>

I have this code in a dial plan: exten => _417XX,n,GotoIf($["${CALLERID(num)}" > "SIP/41799"]?notfromlocal) exten => _417XX,n,GotoIf($["${CALLERID(num)}" < "SIP/41700"]?notfromlocal) The value of "${CALLERID(num)}" appears to be "SIP/41712-00000181" -- Executing [41720 at from-internal:5]

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.4 Snom870 FW = 8.7.3.19 /8.4.8beta I would like to change the background colours on the BLF vkey field based on the station status. I posted the following to the Snom support forums some days back and have had no response so I am asking here in the hope than one or more of you have done something like this: I have the

Hi, Watching Snom 870's video (http://www.youtube.com/watch?v=9e8hPxX0oDU), you can see a new sidecar (phone extension) which seem very interesting. Has someone details on this extension ? Any release date or/and data sheet ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Thu, March 5, 2015 09:56, Ruben R?gels wrote: > > Hi again, > > I'm glad to hear that I provided a somehow useful answer. > > Unfortunatelly, I don't know these details. > If you wasn't lucky consulting the snom docs, maybe the snom support > can be helpful with information about the exact implementation > details. > > You also could use "sip

We have a FreePBX-12 / Asterisk-12 setup that supports about 24 extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence. The server host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x) and is up-to-date. Registrations require very long random passwords and registrable devices are further restricted by

You have to consider whether you really want "anonymous" calls, or you just want to enable SIP calls from trusted companies/partners. The latter means setting up routes to these companies and (ideally) registration between peers. If you really want anonymous calls, then you will have to setup your dialplan with a guest/anonymous context for the calls to drop into. Once they arrive in

On Thu, March 5, 2015 05:30, Ruben R?gels wrote: > > > Am 05.03.2015 um 01:09 schrieb James B. Byrne: >> I am trying to determine how the transfer button on the Snom-870 >> works >> with Asterisk. Is the ## special code employed as when it is >> entered >> through the handset or is the blind transfer through the phone >> function accomplished in a