Hi, we are a small startup from Poland. We are looking for someone who will help us with Asterisk integration. We have our own application based on Laravel Framework. We want to improve our app about dialer features most calling, receiving calls and recording calls. Do you have any experience of that type of work which could help us?
need to enable call recording on asterisk service for 20 seconds to match available recording so the can take further step
Dear Sir or Madam, I am searching for an expert in VoIP and Asterisk or alternative program. What ist he problem/task? I am going to create an android app like „Marcophono“, which you can find in the german playstore. With this app Users can do prank calls. It works like this: 1)User A choose a scenario in the App. 2)In this scenario User A can see different Audio Messages. 3)User A calls User B with the App. 4)If user B answer the call, User A can press different buttons in the App like „Hello my friend“, „How are you“ etc. User B hear that messages directly live, but not the voice from User A. Important: User B did not have this app installed on his phone. Thats the point I need your experience. I think the owner of „Marcophono“ did ...
i need a php script which connects to ami asterisk (this is already done) then it needs to set dynamic agents to be remove or added to/from a specific queue and also we need a shell script which logs off all agents from a specific queue
i need a php script which connects to ami asterisk (this is already done) then it needs to set dynamic agents to be remove or added to a specific queue then we need some jquery tool which polls from ami and displayes if user is logged in to queue or logged off. and also we need a shell script which logs off all agents from a specific queue
We are looking for some Asterisk work on a number of items. We are not using freepbx or other GUI. The current tasks consist of the following: The configuration is two SIP trunks from Twillio. One trunk works the other does not. Need to troubleshoot. All calls coming into the default extension of either of the trunks must simul ring to all extensions existing now and any that are added in the future. On one trunk this is configured for existing extensions only. Needs to work on newly added extensions as well as second trunk. Notification of all missed calls to default extension must be sent to a specified mobile number via text. Any voice mail message left to that extension must also be sent via text and email. Notification of all missed calls to other extensions must be s...
Task - Enable DTMF ( press 1 to connect) if pressed then forward to an external number and if no response disconnect the call.
Asterisk Freepbx Admin - Zoho phone bridge Integration
Hi Ram i have two asterisk conmected via IAX, when call is NO ANSWER or BUSY or CANCEL im not getting good Sip cause but 503, exemple client send call he get ringing 180 than after timeout (no answer) he get 503 instead of 408, the gateway is sending correct sip cause to 1st asterisk but always congestion(503) except when call Answered I think something wrong in or or in my My Architecture is CLIENT --SIP--> ASTERISK --IAX--> ASTERISK --SIP--> GW
Ura com reconhecimento de voz IA para asterisk. Text-to-speech e reconhecimento de voz.
When making a outbound call from Skype for Business softphone unable to hear the ringing on SFB app. But the call rings at the other end and when answer the call works fine. Current setup is when calling from SFB has a trunk connecting to Asterisk freepbx and the SIP trunk to the provider is configured in Asterisk. it would be great to have someone who has knowledge in both Asterisk and SFB server.
Interconnection between WINDOWS SOFTSWITCH that has PUBLIC IP, and Asterisk PBX (DYNAMIC IP),, We want to connect the Asterisk PBX as a trunk to the windows softswitch so calls from Softswitch to Asterisk PBX can be successfully connected.. Please note: DYNAMIC DNS is not preferred.. Thank you.
I need to register my ASterisk PBX to my Softiswtch,, my asterisk PBX has dynamic IP
Asterisk Wallboard. Code php 5.6 Same layout as image below Management portal to control Wallboard Queues to show and which Agents by Queue to show and other settings as colour etc Able to setup more than one wallboard. Able to assign colours to Queues or Queue Groups Lookup Pause name and Agent name from another MySQL DB Only show top queue box if there is a Call waiting Colour change on Queue Box if over 5 calls or 10 calls waiting. (Yellow, Red) Colour change on Agent if over 20 minutes talk time and 40 minutes talk time. Allocate sort order to Queues or Queue groups. Agent last call record is the last call from any Queue
i want to change the path of my extension file in asterisk
I need someone to alter some images. Change some numbers on a Instagram print screen. Top number should be instead of 2.261 267 Second number (actions taken) 524 instead of 108 Replies 67 Profile visits 89 Sticker tap (same number as @_melakitchen_ ) should be 321 instead of 97 Accounts reached 3.577 Impressions 2.491 Please use same font as original Instagram numbers
I have a call center software I need a vici dial asterisk server expert who can change my system layout & make some systems to work properly according to my requirements
Hi, we are looking anyone who can create web interface for asterisk where we done below task 1. add / remove sip trunk 2. check sip trunk status 3. realtime calling (live calls) 4. force/terminate calls 5. generate calls from mysql and send CDR to any Mysql 6. generate TTS calls from mysql and send CDR to mysql 7. Capture User Input (DTMF) and send to mysql along with CDR thanks
Asterisk, PHP, agi, xml call Task description We need information from a web service, from an XML Soap Call. We need to be able to call this php-agi script from asterisk, passing it variables such as telephone numbers or account numbers . XML Calls (SOAP) There are three calls from XML that we would like to call. These are real links 3.PayByCreditCard_New
Hi, I m looking for asterisk devoloper for api integration. We have a IPPBX based upon asterisk and we want IF customer INPUT through DTMF, It should read balance and all.
Install and hook up VOIP phone server (asterisk) Users are recognized by registered phone numbers in profile. Ability to record product name & price (2 separate fields) for each product. Main menu: - Sale menu: leaf through all available products for current sale, select products and add {quantity} to cart. - Existing order: leaf through all items in cart one by one (Press x to , with quantity and price (per item). Ability to edit item order (add / reduce quantity, remove from cart). - Automatically leaf through all order items in sequence. (no editing) - Hear order status message (pending payment, processed, shipped etc.) Previous experience is a must.
Need load balancing setup for apache and asterisk on a multi server ViCiDial setup. We have 1 recording server + 3 SQL server + 6 Asterisk nodes SQL balancing is already set up and all nodes operable. Should be a 15 min job for someone with experience.
Fix issues with caller id issues in asterisk and SFB Fix issues with inbound calls not connecting to voice mail in SFB Setup multi tenant in Asterisk Setup auto provisioning for Yealink and Polycom phones to work with Asterisk
I need to return variables between perl to asterisk pbx ivr. Also parse an xml
Hi Hope you are doing well I have seen your profile seems like you are a asterisk devloper i'm also new at this world So what I need is that : I have a custom crm devlopped by Symfony 2 I have a page that show some clients with their number recently I have install grandstream UCM IP PX for to make a call center i need an script to make a simple call from extension to a custom number (of client ) to integrate it in our custom crm so the agent can click on this number then the call make automatically I have create a script with native php but the problem is that agent has to confirm the call by answer then the call is made like he need to passe by incomming call to lunch the call (this for outbound call (extension to cell phone )) I have try to solve the problem usin...
Hi Hope you are doing well I have seen your profile seems like you are a asterisk devloper i'm also new at this world So what I need is that : I have a custom crm devlopped by Symfony 2 I have a page that show some clients with their number recently I have install grandstream UCM IP PX for to make a call center i need an script to make a simple call from extension to a custom number (of client ) to integrate it in our custom crm so the agent can click on this number then the call make automatically I have create a script with native php but the problem is that agent has to confirm the call by answer then the call is made like he need to passe by incomming call to lunch the call (this for outbound call (extension to cell phone )) I have try to solve the problem usin...
Hi Hope you are doing well I have seen your profile seems like you are a asterisk devloper i'm also new at this world So what I need is that : I have a custom crm devlopped by Symfony 2 I have a page that show some clients with their number recently I have install grandstream UCM IP PX for to make a call center i need an script to make a simple call from extension to a custom number (of client ) to integrate it in our custom crm so the agent can click on this number then the call make automatically I have create a script with native php but the problem is that agent has to confirm the call by answer then the call is made like he need to passe by incomming call to lunch the call (this for outbound call (extension to cell phone )) I have try to solve the problem usin...
Hi Hope you are doing well I have seen your profile seems like you are a asterisk devloper i'm also new at this world So what I need is that : I have a custom crm devlopped by Symfony 2 I have a page that show some clients with their number recently I have install grandstream UCM IP PX for to make a call center i need an script to make a simple call from extension to a custom number (of client ) to integrate it in our custom crm so the agent can click on this number then the call make automatically I have create a script with native php but the problem is that agent has to confirm the call by answer then the call is made like he need to passe by incomming call to lunch the call (this for outbound call (extension to cell phone )) I have try to solve the problem usin...
Hi, I need assistance with a voip migration project, you must have prior experience with 1) Migration from Asterisk to freeswtich 3.6 or 4.0 Multitenant enviroment 2) Setting up FreeSBC / Pro SBC 3) Dialplan setup 4) SIP Trunk setup. 5) Virtualized setup on Vmware or AWS. Project must be completed in 7-10 days.
We have already configured asterisk freepbx on premise integrated with dinstar gsm 8 sim gateway. It is also integrated with vtiger crm. We want to rehost and reconfigure it on our cloud server. We also want few email and sms integrations to be done with overall security check. We need someone who can do all above.
We have already configured asterisk freepbx on premise integrated with dinstar gsm 8 sim gateway. It is also integrated with vtiger crm. We want to rehost and reconfigure it on our cloud server. We also want few email and sms integrations to be done with overall security check. We need someone who can do all above.
Hello, We have a project in Asterisk, and need urgent support,, please message me
OpenVPN server setup needed on VPS for the purpose of remote maintenance. Remote servers (Freepbx/Asterisk) start OpenVPN on bootup and stay connected for the purpose of remote maintenance. When needed, support person would connect to VPN to perform changes.
We need to create SIP compatible softphone for our asterisk server need to include: push notification transfer and conference BLF buttons if possible auto provision template
I need a call flooding script to run on Free Pbx/asterisk simple script not complicated.
-we need this IVR system for a Radio and TV station. -after hearing the menus call will get connect to an extension. -we need reports and WAV recording. -CRM
hello sir i have a projet of voip in morocco but this period i have a problem in my system based on asterisk+A2belling ;i must do a filter to block any number enter more 12 digits and less numbers called number robotic or test number
We are looking for an asterisk / full stack developer who can help us build a call center management platform platform and dialer using Asterisk. Here is the basic outline of the project. This project will require the following. Agent Control (Manage agent and customer interactions) Call Routing Queue monitoring and reporting Provide call information (referred to as call pop-up) to agents before handing over the call 500 outbound channels sip (spanability) administrator gui phone list builder dnc database and scrubbing tool statistics page wav file uploader and player for outbound email for call results .wav file data distributing leads based on availability of agents to multiple customers the completed project must be distributable. If you have built something similar, pl...
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or ...
Hello! I am looking for an Asterisk (VoIP + SMS) developer. I have a Dinstar Device with 8 GSM SIM Cards that can accept and make GSM calls. I need those calls forwarded via VoIP (preferably Asterisks) and distributed equally among Call Center Agents that are currently logged in. This device has a capability of Sending and Receiving SMS, and we need that feature too. A software was already made for me, and it is working great and is hosted in Digital Ocean! However, I need more features, such as reports, scalability, long SMS thread, among others. However, I can no longer contact the developer. He is no longer logging in to his Skype for more than a year! Thank you! ~mark
I build Vtiger server with multiple databases for each organization. The connection to the relevant database set with the login parameters. In addition, there are multiple asterisk servers each is multi-tenant providing services to multiple organizations. I need to connect the Vtiger to the asterisk (the asterisk server ip set in the vtiger relevant DB parameter). The asterisk AMI connection is as standard asterisk server except each extension as his relevant organization as well. In addition, there is a node.js server installed that already connected to all servers and extract all events.
I NEED A SOLUTION TO FIX THIS PROBLEM. " no RBT, Fas voice recording with charge "
I need assistance in setting up my FreePBX home server. I'm using Cisco WebEx Teams and want to use the call in feature using the SIP URI provided by Cisco (example sip URI: <extension>@). So I've set up an extension with the DIAL property pointing to that specific SIP URI (which means that I'm not dialing the SIP URI from an internal endpoint). The problem is that Asterisk is using UDP and looking for _sip._udp. SRV records but WebEx is only providing TLS transport. Now I don't know how I should set this up in order to force TLS for this specific external SiP URI.
I am using call center on Asterisk there are library with and When hold and resume, can not process this event. Web phone built with angularjs Please help me.
I am an experienced PHP coder. I have built 2 major projects using Twilio voice and sms features. I have another client that would like a phone system created. I would like to explore my options with Asterisk. I would like help and guidance from someone with a lot of experience in this field to help me choose the best option and set up the proper platform. If this goes well, then it may be worth it to migrate my current twilio projects to Asterisk or another platform as well. I am looking for someone who speaks English as a first language and someone who can be available working hours of EST.
