Nagios asteriskJobb
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 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.
Especialista em Firewall, Switches, Servidores Windows Server, para contrato de prestação de serviços Nível 1 Avançado! Tenho um prestador de serviços, que estamos prestes a encerrar o contrato e precisarei de outra empresa/profissional para assumir! É preciso ser de São Paulo, para suporte in-loco caso necessário! Especialidades necessárias: Linux Wi...Firewall, Switches, Servidores Windows Server, para contrato de prestação de serviços Nível 1 Avançado! Tenho um prestador de serviços, que estamos prestes a encerrar o contrato e precisarei de outra empresa/profissional para assumir! É preciso ser de São Paulo, para suporte in-loco caso necessário! Espe...
Buenos días, Queremos integrar Asterisk 13 (con freepbx 14 y por debajo Centos 7) con ZOHO CRM. Los usuarios trabajarían desde PCs con windows y softphone instalados. Muchas gracias saludos
I use a production voip server. In past we had customized cid superfecta module (mainly send to growl sub-module) to match caller id with remote database and...server. In past we had customized cid superfecta module (mainly send to growl sub-module) to match caller id with remote database and send us caller details via growl desktop pop-up. this is very easy for php developer After last few system modules updates, this is not working properly (we have only caller details, nor called line info), any php developer can fix this, no need to have voip or asterisk skils. plz contact me for additional info (here : desktop growl popup notifications look - 2 files which have been edited last time - budget fixed : 80 $ in one day - bid only if you re interested and ready to start withi...
Hi, I'm buying existing code or script in various theme. It has to be fast, modern, bug free. Anything can be interesting but the following preferred: Saas, CRM, Real estate, video chat, booking system, asterisk, webrtc, etc. CODE MUST BE ALREADY DEVELOPED AND YOU HAVE TO OWN THE CODE (!!) and connected website has to be presentable with demo back end. I do not care codecanyon links or "I can do it for you OR I'm good in x or y program language". Thank you for understanding - I have very bad experience with auto bids and fake promises. Cheers
We are trying to fix the one way audio issue on Freepbx
hello, i have working asterisk fresh installed server. i need to add my voip trunk into it + my dailplan need Voip expert on urgent complete this configuration
i would like to create a dial plan and the workflow will be as followed there is 1 DID when this did is dialled it will connect to an asterisk server... this will ask for the caller to enter there pin number .. if the pin number exists it will then ask for them to set a caller id ... then it will ask them to enter there destination number which they will be bridged to
...detail is an absolute must. Issues must absolutely be resolved within a give SLA's timeline and constraints, follow ups, KPI's, resolution rates, customer satisfaction and other factors do play a significant role. You will be working with Exchange, Active Directory, Windows 7, 8, 10, Windows Server (virtual and physical(HP, Dell, Lenovo)), Linux (Debian & CentOS), MS-SQL, PostgreSQL, MySQL, PBX (Asterisk, Panasonic, Fortivoice, Mitel, Northstar, FreePBX, Elastics), Firewalls & Switches (Fortigate, Cisco, Checkpoint, Sonicwall). Networking is a must as well, switching and routing (Cisco, Forti, Netgear). As menitoned previously this is a position/contract that will incorporate multiple individuals and when we've mentioned that attention to detail is a mus...
We are looking for a guy who can create SIP based GSM gateway for us by using GSM 800 and QUECTEL M66 module so that we can connect it though asterisk application.
...asterisk16 installed via packages. I want a MINIMAL working asterisk SIP configuration to accomplish these things: 1. Three softphone extensions that can call each other. Two on desktops behind internet firewalls (different networks than the server, with NAT), and one on the FreeBSD server itself. 2. A publicly available extension, like me@, that anyone can use to make a SIP call that connects through the server to a softphone, and lets the caller leave a voice mail if no answer (which would be emailed to me). 3. Everything secure: No unauthorized access. 4. Be able to switch between RTP and SRTP by changing the configuration. 5. If possible: no asterisk database connection, only flat files. If not possible, explain why. 6. Asterisk comes up and runs as cleanly as ...
We're looking for an experienced translator to collect all the website text and translate it from italian to english. This project requires experienc...link leads to an external website and isn't included in the project scope. Please put the very last word of the site's homepage in your proposal as a proof that you've read the job description. Deliverable should be in Excel format, with italian on a column, and english on another. Bold character styling should be preserved, if the document format doesn't allow it, it may be written in **double asterisk** notation. Project doesn't include installation but localization and collection of text will follow two distinct milestone, where we'll perform a check for any missed text (for example, popup di...
we have asterisk 11(freepbx) setup and running. need vtiger install and connect to asterisk.
We are in a requirement of sending SMSBlast/Campaign by using the SIP account. We have VICIDIAL and ASTERISK EPBX. So, by using any of these SIP accounts need to send SMS without gateway.
