similar to: Asterisk Mobile Dialer

Gentleman, Moderators, i don't know if this topic if OFF-Topic, if yes, please tell me. I had some difficult looking for a Asterisk software that provide me some functions (For exemple: CDR, Queue control, MeetMe Control) all-in-one. So i decided to develop than. In a few weeks i'll deploy a Beta version of this software and i'd like to know if is somebody available to try this

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:*

Hello, ? I am interested, too. ? Att, Welinghton Citando Mitul Limbani <mitul at enterux.in>: > Hey Helvio, > > Would like to check it out as well. > > Do email me, > > Mitul > On 22-Jun-2015 9:05 AM, "Helvio Junior" <helvio.listas at gmail.com> wrote: > >> Gentleman, >> >> Moderators, i don't know if this topic

Hi, Try as a first step a tcpdump capture to verify if the softphone is actually sending the register message to the server. For me it seems like the softphone is not able to reach the server ! Best Regards, On Fri, Apr 24, 2015 at 10:55 AM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish

Hi Helvio, Could you tell me what is process to setup an environment for IAX. Regards Akhilesh On Fri, Apr 24, 2015 at 4:25 PM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish > calls. You need to check if there is AWS firewall rule that allow your > communication from your client

Use the command bellow to check if is Asterisk opening the port. netstat -nap | grep 4569 You need to see something like this output, otherwise your asterisk is not opening the port. udp 0 0 0.0.0.0:4569 0.0.0.0:* 10244/asterisk Att, H?lvio Junior dCAA - Digium Certified Asterisk Administrator SafeId - Gest?o de identidades e Acessos +55 41 | 9

Hi Guenther, Thanks for ur reply I have concern from long time I'm not able to login through softphone with AWS Cloud.Please let me know is there any document or guide line for the same. Regards Akhilesh On Fri, Apr 24, 2015 at 1:26 PM, Guenther Boelter <gboelter at gmail.com> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 04/24/2015 03:34 PM, akhilesh

1.8 or higher. Att, H?lvio Junior SafeId - Gest?o de identidades e Acessos +55 41 | 9893-2694, single-sign-on.com.br helvio.junior at safetrend.com.br On 29/06/2015 14:43, Abdul Basit wrote: > 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:

Hi All What, by definition, goes to the cdr table's "dst" column ?? In our setup, to get outside the user has to dial X before any number. This goes to the dst with the X stripped out. I recently made some changes in a macro and after that the X appeared in this dst column! I run through the script and see no differences between the old and the new one. Then the question: What,

Please keep the ?me to? emails off the list. Regards; JV From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Magno Guimar?es Sent: Monday, June 22, 2015 3:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Product CDR/Queue/Meetme Hello, I am interested, too. Att,

Hey Helvio, Would like to check it out as well. Do email me, Mitul On 22-Jun-2015 9:05 AM, "Helvio Junior" <helvio.listas at gmail.com> wrote: > Gentleman, > > Moderators, i don't know if this topic if OFF-Topic, if yes, please tell > me. > > I had some difficult looking for a Asterisk software that provide me some > functions (For exemple: CDR, Queue

Hi Akhilesh, SIP protocol use port 5060 (default) and many other ports to stablish calls. You need to check if there is AWS firewall rule that allow your communication from your client external IP and your AWS host. Also, think in use IAX intead of SIP, because SIP protocol has many trouble when used with NAT, also IAX protocol use only one port (4569) to everything. When i need allow

You can use tcpdump in your server to verify if it is receiving the packets. tcpdump -ni any port 4569 So you have more than one ip in the server? On 5 Jun 2017 9:13 pm, <thelma at sys-concept.com> wrote: > No, I don't think it is IP table issue, I've not upgraded dd-wrt for a > while and it was zoiper was working OK with my previous version of > asterisk. > >

Hello All, I have a requirement to setup a predictive dialer for a customers call center. I am asking for pros and cons of the different dialers available for Asterisk. If you are going to send marketing material send it to my e-mail directly please and not to the list. I was hoping to get the opinions of any one using any of these dialers and what they liked and didn't like, ease of

Hello, We would like to share our webinar about Auto-Dialer Business Model (Telemarketing). It is educational video which we made for our clients and now we are sharing it with you. http://www.kolmisoft.com/how-to-start-a-VoIP-business/webinars/ Enjoy. NOTE: This is not attempt to sell you anything. No product or service is sold/marketed in the video. Regards, Kolmisoft Team

hi wich predicitive dialer are you using and wich one do you recomend? a link to the project/product and a link to a how to will be VERY apreciated. Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all- I have asterisk running on 2 systems, with four E1 spans each. Each system is connected to a (big) NT DMS-100 switch. For load testing an IVR system running on one of the asterisk systems, I'd like to use the other system to generate a lot of outbound calls under program control - on most or all of its channels simultaneously. All of the asterisk dialplan and agi programming

Howdy, I need some pointers (ideas or help) to build a solution to retrieve recordings out of an IVR. The program needs to do some predictive dialing functions I only need it to: 1) Be able on it's own to make a call (to the same number inside this script). 2) Detect that the call has been answered and that it has finished. I need some sort of AGI but it needs to run on it's own, i.e.

Currently I have a predictive dialer that is web-enabled, as well as a superdialer mechanism. I was wondering what kind of success people have had with their dialers (lessons learned, etc.) and how I need some direction on how to seamlessly integrate an inbound system with the predictive dialer. My thoughts are that inbound phone calls will be routed to a receptionist first, who will then

I'm somewhat new to Asterisk and am tasked with having it perform some automated functions. Is there a way with the current system (and/or extra modules out there to:) 1. Launch "something" from a command line (on the Asterisk server) to: 2. Dial an extension 3. Issue some DTMF sequences, 4. Record the output to a WAV (or GSM) file, and 5. exit Any quick pointers would be greatly