similar to: Caller-ID questions and suggestions

Most of the time i can find answers to my questions on the wiki, google, or searching the list now i am stuck . I have a small * box at my house running 1.0.9 stable and a devlite kit. Every thing is awesome VM, IVR, Echo canceling, and Meetme are all working great. But on Incoming caller id i need to add a 9 as a prefix to make it easier to return call from my cordless phone (cheap vtech

Hi, I've been looking at the problem of the default caller ID. When a call comes in with no CID or witheld it's always set to 'asterisk' which is what the phone displays. I've been looking for an option to change that. The only place I can find is DEFAULT_CALLERID in chan_sip.c. This is set by the 'callerid' option in the sip.conf. However the documentation

I am not sure how this works in other countries, but in Australia - according to Telstra - the caller id is managed by the terminating device which is the telephone. If the caller id has been flagged as a silent number then generally the caller id is not display, but rather display with an alternative message 'Number Withheld' or 'Private' or whatever. Right now every

>On Sat, 29 Jan 2005 12:53:11 -0600 > -----Original Message----- >From: <asterisk@draughon.org> >Subject: [Asterisk-Users] RE: Q: Can I over-ride the value of > ${CALLERIDNAME} ? >To: <asterisk-users@lists.digium.com> >Message-ID: <001a01c50633$d9e10a30$6701a8c0@calhoun> >Content-Type: text/plain; charset="us-ascii" > >Folks, > > Many

Hi. I am running asterisk 11 and i have usb 3g dongles to make my gsm calls with the following config in extensions.conf exten => _9X.,1,Dial(Dongle/dongle800/${EXTEN:1},120,KT) exten => _9X.,n,Hangup(${HANGUPCAUSE}) By dialing 9 it opens the dongle to make a call. I would like to restrict my caller id. so when i place a call from this dongle, it will send on the other end *blocked number*

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>

Hi, I'm in the process of switching over to Asterisk from Alchemy kit and have hit a stumbling block. We're in the UK and use ISDN. At the moment we don't accept calls from withheld numbers (we just play them a message), but do accept calls from unavailable numbers. There doesn't seem to be any way for me to differentiate between the two number types in Asterisk (chan_CAPI) -

Hi all I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example in sip.conf: register => andy@sip.mydomain1.org/1000 register => andy@sip.mydomain2.org/1000

On a BT line, how do I determine whether the number on an incoming call has been deliberately withheld (by dialling 141) or is merely unavailable (e.g. because it originated from overseas or passed through some ancient switching equipment) ? In the first case, I want the caller to be played a message to the effect that we are not at home to anonymous cowards but if their business is

Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with

Hi,   I'm using lars package to run some regression analysis and my doubt now is how can I predict my model to another dataset? Let me explain a little better: I have a dataset from which I withhold some data. With the data that wasn't withheld, I create the model. Now, what I'm not being able to do is apply the model back to the data that I withheld. Any suggestions?   Here it goes

Hello, I have a rather odd problem with Asterisk detecting faxes. I have two POTS lines coming into the box (TDM400P). Line 1 is for voice, Line 2 is fof fax. When I set them up with channel => 1-2 in zapata.conf, all is fine, but as soon as I have two channel => definitions, Asterisk is unable to detect faxes. The fax line is not supposed to ring local phones, so the most obvious

Hello, we have a Samba 4 AD with cups as printing server. Windows clients can download driver via samba and nearly all is fine. Expect the cups log on http://dc1:631/printers/Printername?which_jobs=completed Name is always "unkown" and Owner is always "withheld" is there a way to log windows username how print the job? Best Regards,

Hi, now I've got caller id working on my BT line in the UK, I'd like to play a different message to those pesky sort who with hold their outgoing number. How can I do this in my extensions.conf for my [incoming-analog] context? I realise some people may call who are unable to change the way that their system withholds the outbound number, so I'll give them chance to leave a voice

thanks a lot for the reply. i thought of that and i did try to send *exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)exten => _9X.,n,Hangup(${HANGUPCAUSE})* but the provider replies back that it is a wrong number. Then i inserted the sim to an ordinary mobile phone and dialed #31# and the number, then the call progressed fine and it restricted the number. What am i doing wrong

Colleagues: I have encountered the following situation: SERIES OF COMMANDS source("File1") MORE COMMANDS source("File2") Optimally, I would like File1 and File2 to be merged into a single file (FileMerged). However, if I wrote the following: SERIES OF COMMANDS source("FileMerged") MORE COMMANDS I encounter an error: the File2 portion of FileMerged

so any ideas, please? On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux <atuxnull at gmail.com> wrote: > after adding the ww: > root at Pbx: /etc/asterisk $ asterisk -rvvv > Asterisk 11.25.3, Copyright (C) 1999 - 2013 D == Using SIP RTP TOS bits > 184 > == Using SIP RTP CoS mark 5 -- Executing > [9211123456 at AllCalls:1] Goto("SIP/500-00000003",

Hi Most of our customers have generic names like "Hospital", so I need to rewrite their caller ID name by looking up the number in a database on the Asterisk server, and rewriting the name such as "Reading Hospital" so that we know who's calling. Any idea if this can be done with Asterisk, and how to do it? Thank you.

Hi All, I'm about to modify my menu to allow for certain callers to bypass my menu entirely, and just ring my phone. I don't need friends hearing my business menu all the time. So, my idea was to use my already enabled callerid asterisk database and lookup the callerid info from it, if it exists consider it a friend and send them to a direct dial context. If the callerid is new, not

>>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT) My suggestion would be to add a pause or two before dialing the phone number exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT) D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel (you can also use 'w' to produce .5 second