similar to: I want to store cdr into database

Hi, I need to make calls to a list of numbers one at a time and once the user pick the phone connects to an IVR where I can get few data, after a call finishes the 2nd number get called and so forth. I'm familiar with Asterisk/Elastix but the Campaign feature on Elastix does not seem to fill this need. I'm now looking GoAutodial & AsterCC. Anyone with an idea to solve this issue I

I have a similar situation here, I want to include TTS in my asterisk IVR system. Could someone give suggestion(s) please, I prefer open-source thanks in advance! Chatila, A. C. P. O. Box 365, Kihesa Street, Njombe, Tanzania. *Mob:* +255 765 154 235 *Whatsapp:* +255 653 258 608 *Website:* chax.me.tz On Thu, Aug 27, 2015 at 9:07 PM, Steve Edwards <asterisk.org at sedwards.com> wrote:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 > On 2/02/2017, at 9:52 pm, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > > <snip> > > but in simple solidarity with everyone who has ever > > been pissed off by a machine-initiated spam marketing phone call at an > > inappropriate moment, I am not going to tell you how to do it. > >

-----BEGIN PGP SIGNED MESSAGE----- Hash: RIPEMD160 hi all, i have two OSX boxes set up for ssh via pubkey auth only. i'm setting up rsync comms for the first time. i have rsyncd running on box A. no-auth rsync from box B to/from box A's rsyncd works as expected. rsync@B to/from rsyncd@A using pubkey-auth'd-ssh trasport: rsync --verbose --stats --recursive -e "ssh -F

I had been using google tts, but it started requiring a captcha for my browser, and via linux I can't access http://translate.google.com/translate_tts?q=test (redirects to captcha) as so, its not reliable On 27 August 2015 at 17:16, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 8/26/15 1:15 PM, Tech Support wrote: > > All; > > I have a customer who is

my postfix setup http://code.google.com/p/appwsgi/wiki/smtp my dovecot setup http://code.google.com/p/appwsgi/wiki/imap root at localhost:~# /etc/postfix/test/root 220 localhost.localdomain ESMTP Postfix (Ubuntu) 250-localhost.localdomain 250-PIPELINING 250-SIZE 10240000 250-VRFY 250-ETRN 250-STARTTLS 250-AUTH PLAIN 250-ENHANCEDSTATUSCODES 250-8BITMIME 250 DSN 235 2.7.0 Authentication successful

Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue) announce = first member => SIP/FF4C119EEBF8-SLS member => SIP/FF9EF375CCFC-SLS member => SIP/13145555555 at callcentric ;Eric's cell member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.I did trace a call and

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI>

I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is what I have so far but isn't working. Can someone help me out? Thanks, [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten

I was wondering if the following was possible: 2 separate incoming contexts. The first will be used when there is a secretary present. The second will be used when there is no secretary. I know that this can be done using includes and specifying the time in which each separate context would be included. However, I would like to be able to switch them from the reception telephone. For

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I

What I want to do is have one phone number for multiple call bridges (meetme) so that first users are prompted for their call bridge ID then their password. exten => 7001,1,Playback(/var/lib/asterisk/sounds/if-you-know-the-extension-you want-to-dial-press-that-extension:gsm) exten => 7001,2,set $foo to whatever was entered on the phones keypad exten => 7001,3,Dial($foo,60) Thanks!

I will shortly be replacing a couple of proprietary VPN boxes with a FreeBSD solution. Section 10.10 of the Handbook has a detailed description of how to do this. However I remember a lot of discussion about a year ago about whether the gif interface was necessary to set up VPNs like this or whether it was just a convenience, for "getting the routing right". A number of people said

Hello, I want that people from other servers like ekiga.net can make calls to my users. When I do an "allowguest=no" then people from other domains cannot call me. So I think I need "allowguest=yes". Maybe something like this? ------------- <default> include => users <dialout> include => users exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT) <users>

Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it. As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400). I have no idea what I'm missing. My current understanding is that I need to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd

I am following the instructions in https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am trying to make a call from extension Alice (6001) to extension for Bob (6002). When I make the call, I can hear the ringing on Alice's phone (caller), but Bob's phone (callee) doesn't ring, or show a call coming in from Alice. My setup and environment is as follows: Alice, Bob

Hi all, I'm stuggling with how to present calleds to a specific DDI (DID) with Music on hold whilst the call is hunted around 3 phones, then if not answered within a certain period forwarded to voicemail. So far I've got the queue working and the voicemail but not both together. Ive had a look on the wiki and the archives but can't spot anything that might point me in the right

Happened twice. There might also be a race condition and some bad pointers in chan_local.locals_show. First the segfault. CLI> show locals <unowned> -- 6001@default Segmentation fault (core dumped) [root@mars asterisk]# ll -tr total 22260 [...] Loaded symbols for /usr/lib/asterisk/modules/chan_local.so #0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99 99 mutex.c: No such file

We can use the Dial() command with the 'r' option in order to generate the UK ringtone (as we are UK based the default is UK). How do we generate a USA ringtone for example? I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us' (and calling Progress() beforehand) and although this works for Playtones() the Dial command still continues to play the UK ringtone.

Hello, I seem to be having an issue with voice mail on Asterisk 1.6.2.15 (file storage). Whenever someone leaves a message that is distributed to another box (like VoiceMail(1000&1001&1002,u)), but the VM never gets distributed to the intended recipients. Instead, I get the following in the logs: [Dec 13 11:54:50] NOTICE[15965]: app_voicemail.c:4988 copy_message: Copying message from