similar to: What is my syntax error here?

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 = 2.11.0.2 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]

I am dealing with a FreePBX generated dialplan. I have been following the processing traces attempting to make use of the advice I received here respecting setting a custom ring tone. I have discovered that the context I am using for incoming calls is not used at all during a blind transfer. Thus setting a third ring tone for that situation inside that context is an impossibility. I know now

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>

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

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

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

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4 FreePBX = 2.11.0.4 I am trying to understand flow control in Asterisk dial plans and not having very much luck. I have read the Asterisk book from O'Rielly, or at least those parts I believe might apply, but that has not helped me much on this particular issue. What I wish is to set three distinct ring tones on our Snom phones for

Hello, I work in a hosting company, we have tens of Xen dom0 running just fine, but unfortunately we do have a few that get out of control. Reported behaviour : - dom0 uses more and more memory - no process can be found using that memory - at some point, oom killer kicks in, and kills everything, until even ssh the box becomes hard - when there is really no more process to kill, it crashes