Freeswitch voicexml asterisktrabajos

We need to develop an integration between FreePbx and Laravel able to detect incoming calls. To then open a page with information associated with inbound telephone numbers. The project is to be developed with PAMI and Laravel 7 using events

Hello, I have problem to setup NAT on Mikrotik and Asterisk 16 on Raspberry PI. I have no voice between 2 clients Regards, Michal

Need a docker container, that would run GoIP SMS server. The docker needs to be built using composer, Dockerfile Also need it to be integrated with a working GoIP-1 (1 line GoIP device) The GoIP is already configured and working with Freepbx/Asterisk system

I need a asterisk help .I want to know the wrong password. if any sip user is trying to register with wrong password on my asterisk server so how i can see the wrong password or what password they are using..

Pershendetje Ardit, keni eksperience me Suite crm apo Vtiger crm? kam nevoje pernje konfigurim ne lidhje me crm dhe asterisk

asterisk I need to configuration client said on openwrt (open wrt running on tp link router)

asterisk I need to configuration client said on openwrt (open wrt running on tp link router)

asterisk I need to configuration client said on openwrt (open wrt running on tp link router)

Need to setup dialing script/bot. I have working Freepbx with 1 usb dongle. AUTO DIAL BOT WILL PLAY: Each time when someone pickup have call from BOT, will play 3 steps record. Step-1: Hello, we are calling... Step-2: Please choose 1 or 2... Step-3: Thank you... IMPORTANT Calling from 2 new dongles (sim cards). Can be done in different way. Each time I will upload new dialer list from XML. Can make 2 small list for each dongle. Need current status list (choice1 and choice2),after job is done. Web view or different way. BOT: Each number 2 time try, after no respond again after 3,4h. When some one will call to dongle number - need to rout call to Play 3 Step record again. Extra need button with time set to start BOT DIALING (ex. 9:00 or 12:00) VERY IMPORTANT INFO: I have working FreePBX...

Hello, we need somone to install a webphone on an existing Debian 8 Server that has apache + SSL already installed. Use webphone found here: We have a freeswitch Server installed and ready with test extensions. We need this done today! We are willing to pay a bit more for the rush. So, Please respond to the task with "Dingo I'm Ready" at the beggining of your message so that I know you have read.

We have a single channel SIP account with Skype Connect (Skype Manager) as a trial, more channels will be added. We want to integrate to our asterisk server to make outgoing calls via Skype. Please, we only interested those freelancer who have done it before and not those wanting to experiment. Project is only consider done when we can make calls from SIP phone to PSTN number via Skype Connect Thanks

I am getting error on the asterisk regarding the packet failure Below is the error [16:24, 12/9/2020] +91 98300 52277: han_sip.c:4151 retrans_pkt: Timeout on 588608927-1677186229-42297175 on non-critical invite transaction. [Dec 9 16:24:18] WARNING[1975]: chan_sip.c:4151 retrans_pkt: Timeout on 2000998286-245292724-1360354902 on non-critical invite transaction. [Dec 9 16:24:18] WARNING[1975]: chan_sip.c:4151 retrans_pkt: Timeout on 671697185-364125519-1082815926 on non-critical invite transaction. [Dec 9 16:24:20] WARNING[1975]: chan_sip.c:4151 retrans_pkt: Timeout on 832013417-1702615591-33504755 on non-critical invite transaction. [Dec 9 16:24:20] NOTICE[1975][C-0001ae39]: chan_sip.c:19537 send_check_user_failure_response: Failed to authenticate device <sip:100@>;tag=...

Adaptor needs to work in intercom system (intercom for multi-apartment buildings) Adaptor based on Ruspberry; Receiving analogous signal, and transfering it to digital; Video translation through RTSP; Adaptor registration in Automatical Telephone Station, type Asterisk, 3CX;

Hi, We need a shell script, which runs every 5 minute, it must go into the folder (/var/spool/asterisk/voicemail/default/271283/INBOX/) and check if there are any *.wav files, if there is, then it must rename the files to , if there are more than one, then after "Besked" then it must insert a number, e.g. , VoicemailBesked1.wav... The script must hereafter attach the file in an e-mail and send it to @ The subject of the e-mail must be Ny telefonsvarerbesked fra [telefonnummer] Main mail: Du har modtaget en ny besked på din telefonsvarer fra [telefonnummer] d. [dato], [tid] til [Firma] Lyt til beskeden i vedhæftede fil. Med venlig hilsen NordicCall The above information you will get from the txt file, which is placed in the

Hi I need a freePBX ASTERISK Expert I have some little task which need to be completed as soon as possible I want to make my set up very secure I want the phone to be connected to PBX server trought VPN server Need to fix security certificate and mail SMTP for now

mettre en place un cahier des charge pour un sophtephone ou webphone j'aimerais développer un shotphone, j'ai déjà un serveur Asterisk en place qui fonction et il me manque que le softphone pour mon application

...displayed when requested. Movement commands for the robot will be entered at the console and the floor can be displayed at any time by selecting that option. When the robot reaches an edge, it will wrap around to the opposite edge. Print all results of the robot’s movement to the console when requested. The robot can be placed anywhere on the grid when the application starts. The pen can draw an asterisk ‘*’ or the number 0 (zero). An empty grid is composed of the number symbol ‘#’. You should have options to clear, display, save all robot operations/movement to a file, and read a file containing robot operations/movements. Actions that must be saved(Case does not matter) U pen is up D pen is down <direction>:<integer> spa...

I need a asterisk help i have server with asterisk .I want to see the password of rejected sip user in asterisk..

Im looking for a freeswitch live stream video developer with amazon aws experience with windows docker ec2 and verto

Necessito de uma orçamento para criação de um chat de voz em php baseado na plataforma de telefonia asterisk

I have install Fusionpbx and set gateway (working), extensions, inbound and outbound rules, registered SPA508G phone. Inbound calls work, outbound calls and extension calls all give Temporarily Unavailable. I'm sure it's a quick fix for someone that knows Fusionpbx, I come from asterisk/freepbx, so it's all new for me. I will give teamviewer access to both linux box and Fusion website.

I have to setup a 2000 SIP trunk IVR Bulk Calling setup, Need a experienced Asterisk PHP Linux CentOS developer who have depth knowledge of asterisk and all the IVR Calling setup.

Hi Ajay J., I noticed your profile and would like to offer you my project. We can discuss any details over chat. We can discuss any details over chat. Please let me know if you are available to work on this project. just need to connect vtiger crm with asterisk (both installed on our vps)

Hi Mohammad Abu S., I noticed your profile and would like to offer you my project. We can discuss any details over chat. Please let me know if you are available to work on this project. just need to connect vtiger crm with asterisk (both installed on our vps)

Hi, I have a crm set up and running, Vtiger self hosted on a vps. I have installed asterisk on the server too and need someone to connect and set up asterisk with Vtiger crm. (vtigerasterisk connector is installed too but need support to link it up, or use another alternative all together). This is a very urgent job to be finished!

I am looking for a FreeSwitch live stream video developer with Amazon AWS experience with windows, docker, ec2 and verto

we have a server pc for an asterisk call center but we need to connect the local network and provider network at the same time and also knowledge about freepbx and linux

We are looking for technical solutions for our Asterisk VoIP Server setup with Android as client side that uses pjsip library. VoIP feature in our application connects one user to another similar to WhatsApp calls. We have done the Asterisk server setup but we seek solutions for following: 1. When a user is calling we are unable to identify the callerId for the second user. 2. When the second user is offline we are unable to connect. 3. Require recorded voice messages that are played when another user is busy/offline etc

Configure and maintain a Sangoma SS7 SBC. It has SS7 connections with operators and a Sangoma Freeswitch setup.

Looking to set up a custom skinned rebranded VICIDIAL ISO Image for call centre application . Experience is Asterisk , Vicidial , Free PBX , Linux etc preferred.

...following guidelines: -clever/funny -photo real, not cartoony -homey/cozy feel -shows product (projects) The card “opening” should be achieved either by having an animation or utilizing scrolling in a way that delivers the punchline appropriately. The digital assets for the projects will be delivered to you and we need 2 versions of this “card”, the only difference is that one will have an asterisk in the text with a footnote that reads “gift amount in accordance with EVP reductions” or some such joke they will understand. I have attached an example of the "cozy" element we are looking for that was on a card in years past. This is time sensitive now that we are in Dec., our previous artist was a flake that put us behind schedule....

i need someone who can design a web-based sip phone. which can communicate with any SIP server like an asterisk, freePBX, VOIP, SIP Server, and the browser should support firefox, explorer, and chrome. Also, am expecting a demo which is already done or ready one, so that will agree for implementation

Configure server kamailio with modules : NAT TLS Integration MS Teams Register SIP User Local. Options Forwarding SIP Register to Asterisk Call Routing, OUTGOING. Softphone -> Kamailio( Integration Asterisk) -> MS Tems 365 INBOUND MS Teams -> Kamailio( Integration Asterisk) -> Softphone Softphone -> Kamailio( Integration Asterisk) -> MS Teams 365 Skills: VoIP, Asterisk PBX, Linux

To invoice clients and credit agents with none standard Invoices and Credit-Notes. See the 3 attachments for more information. --- Please note that You must be on this site to answer, because of bugs in the system.

...Dec 04, 2020 We have a project which is a Help Call system for the elderly. It's basically a fancy Voip phone running Openwrt and Asterisk on the MediaTek MT7688 CPU. We are using a WM8960 codec connected via i2S to the CPU and we are performing some basic audio functions to check the audio (record & play) using arecord and aplay. We are getting some odd results such as noise being introduced and fast garbled recording. We suspect issues with the audio driver. It is possible some of the registers on the codec are being messed up. We are looking for some direction as we need to solve the audio driver issues before we get the Voip calls moving through Asterisk. Our first task is simply to play and record audio cleanly using arecord & aplay. Any assistance/d...

We have a requirement to fine tune a VICIAL configuration consisting in the following setup: * HP Proliant DL380 server with 64GB of RAM with a planned upgrade to 128 GB HDD 1.2TB SSD 800GB This hardware has installed VMware 10.7 on it and Asterisk + VICIDIAL instances are being created on it This setup should allow more than 250 simultaneous calls but currently can't handle more than 200 so we need to know where is the problem. In addition we would like someone who can do maintenance remotely via a Windows Desktop in Amazon EC2

Small office with 4 analog incoming lines and 10 extensions. Currently using UCM 6204 on which extensions are setup. Need a solution for accessing logs of: 1- Missed calls 2- Incoming calls 3- Outgoing calls 4- SMS to be sent to missed calls only Be able to access the system from local lan

DetailsProposals Project Details Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File: Range;country;Number;Carrier Payout;Carrier Pay Term;Client Payout;Notes (remarks) -Numbers should be all routed to a local IVR (or a group of IVRs, which ...for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" including Asterisk/FS Skills Required PHP JavaScript Software Architecture ...

Condifurar gateway y modulo call center

To invoice clients for incoming calls and credit agents for work. Please see attached file for more info.

I'm looking for someone who can develop an integration between Asterisk (v.18) and AWS Amazon Polly. Call flow: 1) Asterisk will answer the call 2) Amazon Polly will greet the caller - part of the text is fixed, a small portion is dynamic 3) That's it. There will be no interaction (voicebot). Asterisk will answer, Polly will greet and byebye. This machine is running under Debian.

Preciso de uma api que quando a ligação entrar no queue e ramal que vai atender tirar do gancho ele disparar uma agi.

I have asterisk pbx system and I am looking to update it by adding a phone application. for example, if the phone is not answered in 3 rings it starts ringing in the phone app.

I need a experienced one to develop an Asterisk Open CTI connector with Salesforce CRM. SalesForce Lightning Platform Starter requirements - Enterprise Edition.

...ability to run asterisk 1.4 on the Community Edition of Clearos. the OS can be found (with the MARIADB installed) Asterisk files can be found: https://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.11.1+ Asterisk should be compiled with the following options files ( and ) NOTE: res_crypto MUST be included as a requirement Deliverables must include a step by step reproduce-able solution that compiles and allows the Asterisk to be run from the asterisk -cvvvvg command

Hi, Looking for some one to configure mod_nibblebill in freeswitch to use it for billing. regards

Looking for a professional company that can develop a reseller VoIP platform based on Freeswitch. The user interface has to be unique to us with the ability to integrate into a WordPress website too. Branded Mobile apps, webrtc, Windows and MAC, will also be required. Full feature list will be provided ahead of awarding.

Configure server kamailio with modules : NAT TLS Integration MS Teams Register SIP User Local. Options Forwarding SIP Register to Asterisk Call Routing, OUTGOING. Softphone -> Kamailio( Integration Asterisk) -> MS Tems 365 INBOUND MS Teams -> Kamailio( Integration Asterisk) -> Softphone Softphone -> Kamailio( Integration Asterisk) -> MS Tems 365

I have a CRM created with php codeigniter framwork, I need to start calling via this crm (making outboud calls), so by connecting it to an asterisk system and a sip account.

Hi Viktor S., are you expert in troubleshooting a "toll fraud" attack on our asterisk/freepbx server?