similar to: CallerID setting issue with withheld numbers and mISDN ...

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

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,   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

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

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, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at

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

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*

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",

>>> > 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

Hi All, I'm using PrivacyGuard to filter calls from withheld numbers. A few percent of incoming calls from my BT landline where I know the caller does not withhold their number. BT deny that they're not passing CLI from all calls. In /var/log/asterisk/messages, the following three lines preceed every such call so far: [Jul 11 11:52:28] NOTICE[23198] chan_dahdi.c: Got event 18 (Ring

Hi, I try to compile mISDN 1_1_9_2 (kernel: 2.6.32.13). I had the following compiling error: make[1]: Entering directory `/usr/src/linux-2.6.32.13' CC [M] /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.o /usr/src/mISDN-1_1_9_2/drivers/isdn/hardware/mISDN/sysfs_obj.c: In function ?mISDN_register_sysfs_obj?:

Hi, Until now I've been using my Digium B410P cards with misdn 1.0.x. I would like to upgrade my systems and am now wondering which is the "best" route to take: - use the latest release of misdn v1 - upgrade to the latest "stable" kernel and use the built-in misdn v2 - use misdn v2 as a seperate package (disable misdn in the kernel) - use dahdi's support for misdn

Hi All I have noticed that when calling over mISDN channels the call itself is made over the channels listed in misdn.conf eg: 1,2,3,4 as misdn/1, misdn/2, misdn/3, misdn/4 etc. However, when looking through the CDR's the call gets logged as misdn/0, misdn/1, misdn/2, misdn/3, misdn/4, misdn/5 etc. I also noted that busy, failed, congested calls link mainly to misdn/0. The only logic I

Hi i've installed this software: ******************** SOFTWARE mISDN-1_1_7 mISDNuser-1_1_7 Asterisk-1.4.15 ******************** SOFTWARE misdn is correctly loaded by misdn-inist start Here there is the misdn.conf (copied from an existing and working installation with Asterisk 1.2.x and one BN8S0) ******************** MISDN.CONF [general] misdn_init=/etc/misdn-init.conf debug=0 bridging=no

Hi, I'm installing Asterisk on Ubuntu 6.10 When I first compiled the zaptel package I used: make clean make make install So far so good, but the following command failed: make b410p I did some digging on google and found a guide on how to install it manually, but the result was the same. I got these files ftp.digium.com/pub/telephony/zaptel/b410p/misdn-b410p.tar.gz

Hello I fixed my problem. I changed user and group in /etc/mISDN.conf: <devnode user="asterisk" group="asterisk" mode="644">mISDN</devnode> now it works again! Thanx 4 help! ;) Ingrid -- "Bonnie & Clyde der Postmaster-Szene!" approved by Postfix-God http://wetterstation-pliening.info http://dokuwiki.nausch.org --------------

Hello, I have a strange problem with misdn.conf When I run asterisk, I have this message : Apr 3 23:00:25 WARNING[6824]: misdn_config.c:579 _build_general_config: misdn.conf: "use_callingpres=yes" (section: general) invalid or out of range. Please edit your misdn.conf and then do a "misdn reload". Apr 3 23:00:25 WARNING[6824]: misdn_config.c:579 _build_general_config:

I would like to know if the following misdn warnings are relevant. Currently, I don't need echotraining. However, I took a quick look at the * source code and l1watcher_timeout seems to be defined (echotraining was not found). Currently I'm setting l1watcher_timeout to 0 which is default (so I suppose that this warning won't affect me). Any comments on this? *CLI> misdn reload

Hello, After upgrading my kernel to mISDN-1.1.1 while keeping asterisk-1.2.16 I no longer match any extension. Apparently the "dad" is empty. However I can see the number just before it (146472130): P[ 4] I IND :SETUP oad:!?145201798p ?146472130 dad: ?146472130 pid:2 state:none P[ 4] EXPORT_PID: pid:2 Mar 23 09:35:28 WARNING[6725]: chan_misdn.c:4750 chan_misdn_log: Extension can