similar to: Setting the vkey background colour on Snom870

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>

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:

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

This seems to me to be getting down to some sort of problem with configuring the Snom-870. when I register the device 41712 (set up for transport=tls only) then I see this in the SIP trace: Sent to udp:192.168.6.9:5060 at 4/3/2015 09:07:36:813 (836 bytes): REGISTER sip:voinet09.internal.hamilton.harte-lyne.ca:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.112:5060;branch=z9hG4bK-udx92poqese6;rport

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

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]

>>>>> "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

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>

On Thu, April 9, 2015 12:37, Tafadzwa Nyabasa wrote: > Hi There, > > Does anyone know how to program Snom phones using a Mac addresses like > what > is done with the Ciscos. I have about 50 extensions to be programmed > and I > am hoping there is a way on Asterisk to program extensions on the snom > phones. Please assist. > > Regards > I do not think that this is

Hi After recently upgrading to 1.8.3 I have noticed that the nat setting for my peer in my sip table is not making it into the realtime cache. For example * Name : 501 Realtime peer: Yes, cached Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : pack-local Subscr.Cont. : <Not set> Language : AMA flags :

Hi, We face a strange behavior with Asterisk 1.8.15 and SIP defaultuser definition. in sip.conf [blabla0](natted-phone,ulaw-phone,callgroup1,snom-320) defaultuser=tel-221 mailbox=221 callerid="My CID" dtmfmode=auto ;defaultip=10.0.12.21 CLI sip show peer blabla0 Addr->IP : 10.0.12.21:2067 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1

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