search for: ruffling

I'm trying to keep the DND status of my Snom phones and the astdb in line but I'm stuck on integrating my gui DND button which talks to * using the manager interface (actually it uses Astmanproxy as the gui host is on a different network to asterisk and can't see the Snom's across the network). All's working fine in my Dialplan; when someone dials the code for DND-on or

Hi, R Developers, I'm interested in porting the R example datasets package to GNU Octave and Matlab. Would you have objections to my doing so? This would involve transforming the example data and metadata into a format that Octave understands, and porting all of the datasets' Example code pieces to Octave M-code. (This would require no work on your part; it'd be my project.) I think

Quoth: OCG Technical Support <support at ocg.ca> > >Although we've programmed the softkeys per the manuals, they seem to have no >effect (just dead). For example, our 57i is setup like this: I had similar problems and ended up using the speeddial inband functionality. FWIW, my 57i's setup like so: softkey4 type: speeddial softkey4 label: "*Park" softkey4

Quoth Steve Murphy... >Date: Mon, 12 Jan 2009 08:51:03 -0700 > >QUESTIONS: > >Which of the two would you see being useful to you? Obvious comment really but given LEG based CDR, one can determine the 'Simple' data but you can't work it the other way. I'd therefore find LEG based CDR more useful as the granularity (time on Hold, in Queue, Waiting on pre-xfer ring

I don't mean to ruffle any feathers, but has this been properly resolved? It shows that a patch has been submitted to allow connections over TLS, and that the bug has been closed, but the solution supplied doesn't actually address the problem - running Samba in ads mode in a domain with ldap signing. We do not have the ability to change that setting in the policy. Thank you, Scott

I'd like to be able to generate a Manager Event from the dialplan but can't seem to find a way to do it. Alternatively, trigger and Event when a record in AstDB gets changed. Can anyone point me in the right direction? Thanks. By way of explanation, I've a app that connects to astmanproxy and I'd like it to know when a call group gets put into Nightservice. Putting the call

I have two Asterisk systems that can route to each other via a VPN with firewalls disabled for testing purposes. Each Server can see (tested via nmap) UDP port 5060 on the other. So... I thought that I could simply use a Dial command in Server A's config to place a SIP call to Server B... but it doesn't seem to work. Server A (192.168.1.33) has: exten => *136,1,Dial(SIP/90 at

Is there a SIP header I can set (for Snom and Yealink phones if that's relevant) or any other mechanism to tell a phone to ignore a particular call from it's missed call list? I have bits of the dialplan that ring groups of phones eg: exten => 200,1,Dial(Sip/112&SIP/113&SIP/114) and I don't want such calls being recorded by the phone as a missed call. Calls to the

cdr_mysql doesn't set the userfield when it's set inside a macro called from a feature (1.4.25, addons 1.4.8). I have a feature code: autorecord => *1,self,Macro,apprecord The apprecord macro looks like: [macro-apprecord] exten => s,1,Playback(beep) exten =>

I've just started deploying these (well the T28P model) after years of Snom issues and they look pretty good (although the documentation is execrable; if you thought the Snom stuff was obtuse Yealink have got them knocked into a cocked hat!). Anyway, for provisioning I use HTTP with a DHCP entry like:- # # Yealink Phones # group { #

Quoth Jon Morgan <jon.morgan at motors.co.uk> > >We have a 2 port Digium TE220P card, one span is configured to connect to our ISDN30 provider (British Telecom), the other span connects to our internal PBX. Here's the zaptel.conf snip: > >span=1,1,0,ccs,hdb3,crc4 >bchan=1-15 >dchan=16 >bchan=17-31 > >span=2,0,0,ccs,hdb3,crc4 >bchan=32-46 >dchan=47

Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Something with (potentially) lots of BLFs, large(ish) screen, headset and most importantly the ability to transfer calls? I've installed five Snom 370s that seemed ideal but my client is very very unhappy as the Snom 370 can't transfer a call correctly if there's another call coming in (details

Hello Happy New Year to all!! I've just completed porting from Asterisk 1.2 to 1.4. I did this by doing a clean install on a new box, and moving over configuration and scripts where needed. All went surprisingly well! Anyway, one lingering issue is that the function key "lamps" on our Snom phones have all stopped working! We're using a mix of Snom 290/320/360 phones and

Here is my setup 7960(A)--Firewall/PAT--dsl---------Internet--------dsl--Firewall/NAT---7960(B) | | | | 7960(C)--NAT--cable----------------- -----dsl -- Asterisk (A) can communicate with (C) only when C is configured with canreinvite=no. The call gets dropped in few seconds if canreinvite is set to yes for C.

Hi all, I know I'm probably stirring up a hornet's nest with this question/comment but I've spent the last few days working on a PHP-based class for the manager interface as we're preparing for a pretty big upgrade at our call center and I'm revamping all of the management apps I've written. I can connect to the manager interface and send query strings back and forth all

This is mildly off topic since it has more to do with TCP/IP networking than with CentOS per se. However, at the risk of ruffling a few feathers I would like some advice on how to resolve the following routing problem: Given: dual homed host running CentOS-5.2 with eth0 = 2xx.yyy.zzz.23 eth1 = 192.168.219.1 and a Cisco gateway at 2xx.1xx.y7y.1 with the following networks: interface FastEthernet0/1 description connected to...

Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough. I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover cable). This PBX used to be able to

Can anyone help with the use of regexten? (* 1.4.3) I've got Asterisk creating extensions for my SIP phones using regexten but I can't seem to figure out how to make use of them once they're registered. Here's my dialplan for from-sip (the SIP's default context): asterisk*CLI> dialplan show from-sip [ Context 'from-sip' created by 'pbx_config' ]

On 5/6/19 7:47 AM, Dirk Eddelbuettel wrote: > > On 5 May 2019 at 10:47, Andrew Janke wrote: > | I'm interested in porting the R example datasets package to GNU Octave > | and Matlab. Would you have objections to my doing so? > > You don't even have to ask... > > [...] > > | Since R's datasets package is GPL, I think I'd be within my rights to > |

Hi All, Has anyone hooked up * as an extension/trunk of an Avaya system that has around 2 ISDN30e's. Trying to add 100 extensions to one of our systems, but not sure where to start reading. Thanks. -- Kind Regards, Gavin Henry.