similar to: - Asterisk+csv+auto dial out+deliver message

Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the

Following new features are now supported by asterisk based telemarketing software Auto subscription / registration after call recipient press a key in voice broadcasting https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer There will be restriction to call a number in off time accordingly to timezone of

All; I have an application that dials a list of numbers and then plays a recorded message. My customer uses it to dial a list of customers to confirm their appointment for the next day. No biggie, maybe 25 - 30 calls per day for customers who want the confirmation call. What they need now is a way to dial an extension after the number is dialed and answered. I've seen that before, but I

Not sure if anybody noticed, but it seems like Skype For Asterisk is out. $66 per channels, pretty pricey http://store.digium.com/productview.php?product_code=1SFA0001 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090817/cd8c6546/attachment.htm

Hello, I'm currently developing a simple asterisk application using SFS (Skype For SIP) which tries to call to an outbound number, play a message and read DMTF digits. My first approach used the Manager to originate calls and then called an agi script to deal with the rest. Anyway, this ended up being not so clear because the call did not start on the Originate extension that it was supposed

Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human

-----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. > >

I am wondering what the best solution is for initiating a call from Outlook Contacts. I imagine something that would start the call from the outlook card (or similar) and then dial the user's extension and the contact's phone number and place them in a bridge. Anyone use something like this? Travis Ryan Director of Information Technologies Oscar Winski Company 2407 North Ninth Street

Not really, doing the way below you don't even have to worry about it. They both go out at the same instant and as soon as it hits voicemail it disconnects the other leg. If you wanted you could leave it ringing for twenty minutes and it would still have the same effect. Kind regards, Matt > On Feb 6, 2017, at 12:29 PM, Tech Support <asterisk at voipbusiness.us> wrote: >

I'm in the process of writing a simple autodialer to dial a list of numbers and play a message. One of the options I want to give them is a way to "dial X to have a customer service representative call you" Looking for a simple way to pass the number that I dialed to a script in extensions.conf... something like this: [serviceinterruption] exten =>

> We once developed a reminder system for a customer. He's a cleaning > company, cleaning homes and offices. He was spending two hours a day calling > his customers to remind them of their appointment the next day. Two hours a > day equates to 40 hours a month that he saved with that system. He's been > using it for maybe 6-7 years now and not once was a customer upset

Hey; What happens is that a script logs into the AMI and originates a call. When the call is answered, it jumps to a context in the dial plan. Thanks Much; John V. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Victor Villarreal Sent: Monday, May 15, 2017 12:08 PM To: Asterisk Users Mailing List - Non-Commercial

All; I'm currently running Asterisk 1.8.15-cert7 and am using realtime to store my voicemail configuration. The voicemail application works fine, but the problem I have is that the 'Directory' app cannot find any entries because there are no entries in the voicemail.conf file. When I add a context and an extension entry in voicemail.conf, it works the way it should. Is there

Hi Helviom I am interested to evaluate your product. What asterisk version you build this product around? -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On Tue, Jun 23, 2015 at 7:34 PM, Tech Support <asterisk at voipbusiness.us> wrote: > Please keep the ?me to? emails off the list. > > Regards; > > JV > > > > *From:*

Hey; Thank you very much. I was able to install asterisk from your link. One other question. How are you starting asterisk? Do you use an init script or systemd? Do you think that you could share the script you use? Thanks Again; John V. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent:

All; I am trying to build and install certified Asterisk 13.13 cert3 on a Ubuntu 16.04.2 LTS host without much success. I am getting the following errors when I try to compile. [CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o res_pjsip/config_transport.c: In function 'transport_apply': res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg

I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just

On Monday 19 June 2017 at 18:12:35, Sebastian Gutierrez wrote: > use replication 1. Agreed - use replication. 2. If you want an HA (High Availability, not dependent on a single Master DB server replicating to a slave) solution, consider setting up Master-Master replication, with an LVS (Linux Virtual Server) HA machine in front of the two, so that writes can go to either server using only

If I correctly understand what the problem is, what I did was write a script that runs out of CRON every 15 minutes. It checks the outside IP address by querying http://checkip.dyndns.org and compares it to the IP address stored in the parameter ?externip? in the [general] section of sip.conf. If the two values are the same, the script exits quietly. If they are different, the script updates

hello everybody i want to send fax via asterisk in pass through mode. everything is ok if enable fax detection in ooh323 and write fax extension in extensions.conf file. just one problem: delay. i have to wait 5 seconds in order to fax detection done. it is too long for me when i have voice call and no fax. my phone rings after five seconds. is there any way to omit or reduce this time? i test