Openser asterisk jabber jobs
Solve Asterisk Record problem - Record (,5,40,xk) recorda an empty file. We will give you access and you you help us resolve the issue.
Hi, i am trying to set up a basic asterisk, everything is up and running, but i am not able to do outgoing calls. My provider tells me to put the outgoing phone number i want to use as contact in the invite header, but i am not able to modify it, dont know if it is becuase i am running pjsip. Do you think this is something you can help me?
Hi Mohammed. i am trying to set up a basic asterisk, everything is up and running, but i am not able to do outgoing calls. My provider tells me to put the outgoing phone number i want to use as contact in the invite header, but i am not able to modify it, dont know if it is becuase i am running pjsip. Do you think this is something you can help me?
Hi Ambiorix R. i am trying to set up a basic asterisk, everything is up and running, but i am not able to do outgoing calls. My provider tells me to put the outgoing phone number i want to use as contact in the invite header, but i am not able to modify it, dont know if it is becuase i am running pjsip. Do you think this is something you can help me?
We need a person to help us with Vicidial/asterisk configuration. We want to upload a list, do automatic calls which will say: "Say yes to connect to an agent". If the person says "YES" the call is forwarded to agent's phone (through SIP/asterisk)
We are hiring a person to build and configure a asterisk project from zero. We are building a call center.
Hi i need one to install Asterisk freepbx on my server and do auto dailer to dail a special number by sip send call to ip i can set how call can done and how long of call and sperate time between each call maximum calls in same time 5 calls also set ivr on same server if any one from out side send call to ex 555 play ivr my own audio
We are looking for solution like a traditional GSM or CDMA VoiP gateway. This project will be separated in two par...Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP protocol. Certain number of registered Mobile will be able to assign in a group of gateway. On server have to include option to show balance th...
We are using Bicom PBXware MT system and we are looking for someone to help our customers with tech support on live chat and phone, the system is very easy to learn and perfect English is required along with some experience with Asterisk/Hosted PBX
Hi usmanshery, I noticed your profile and would like to offer you my project. I would like to implement a C wrapper for Asterisk AGI functions. If all goes well, there may be more work to do as well. It is intended to implement several functions for an AGI interfacing with an Asterisk process.
To create a dialplan based on specified criteria for incoming calls to server. To add menu option on web interface. Offers between 100-150$.
Hi All, kinldy we need to hire a team to 1- Implement Media Server using UniMRCP 2- Integrate a TTS (text to speach) with UCCE (Unified Cisco Communication Entrprise) and Asterisk 3- Integrate a ASR (automatic speech recogintion) with UCCE (Unified Cisco Communication Entrprise) and Asterisk Thank you.
...problem with server setup for Asterisk integration with Google API. We need to solve that problem. We started the project: By default, the project deploys all the necessary components, including asterisk 13. The only change we made is the channel driver, instead of pjsip we use sip. In the asterisk configuration files, we put the service key with the role of the DialogFlow API for connecting to DialogFlow services. The project contains this library Here you can compile a test client, in which if you insert the above key, they execute the DialogFlow requests. In asterisk, an error occurs during a call, presumably related to receiving authorization data: -- Executing [asterisk-irgkin@dialogflow-loop:2]
Hi, I would like to be able to write in freepbx outbound route (we have few) several numbers as optional caller-id's and the system will choose one randomly for each call using that route
i have a asterisk openvpn server and my client connect server to puppy linux, i need develop my client system, puppy linux exchange TP-LINK router running OpenWRT, this vpn for voip business,
Need a linux expert to secure my goautodial server from DDos and asterisk attacks. I'm using goautodial 3.3 installed on centos 5.
I need a asterisk professional who already have the system ready.
We have installed open source VoIP platform Asterisk on our server. It's need to install to Asterisk speech driver for Google Dialogflow. It's need to connect simple Dialogflow bot to the Asterisk. Here is the driver code:
Setup my asterisk server for allowing WebRTC video calls
Hello. We need a person to help us with Vicidial/asterisk. Calls are not getting through, extensions are working and internal calls are working but inbound and outbound calls are not. SIP trunk is registered OK
Hello! We are a small fintech company based in Ukraine, bringing Fintech B2B solutions to global customers. As a top provider of B2B services, we help our customers make the decisions and follow the paths they need. With Blockchain, cry...especially situated to help our customers with their biggest business challenges. R&D team is concentrated in Kiev, and the company serves clients worldwide. We do qualitative software products to enable the business becoming more efficient and successful. We are looking for local VoIP providers or companies in Arabic region to start selling our products and provide customer support service. We use Asterisk (FreePBX) IP-PBX software calling huge amount of numbers worldwide. We will be more than happy to start cooperating with VoIP companies...
Hi i have Asterisk installed on raspberry pi all i want i want to allow calls from one public voip ip server with prefix go to extensions regards
Hello, We looking for freelancer who is expert in integration stuffs like twilio with this opensource java CRM - We need to also install twilio dialer and Asterisk server So we can achive the predictive dialer and auto dialer and SMS, whatsappp messaging - with two way communication, with the twilio integration as service provider. BId only if you have this all idea and can do fast. Thanks!
You need to write or provide a connector for integrating FreePBX 14 based on Asterisk 13 and Vtiger 7.1. It is also required to provide detailed instructions for installing and configuring the connector; after installation, Vtiger users must be able to make outgoing calls, receive calls, and listen to them.
Hi Yuriy M., I noticed your profile and would like to offer you my project. We can discuss any details over chat, i need you install for asterisk and free pbx or other solution for usb dongles for voip termination could u work on it and will need support as well after the project so what do you think ?
salam kamel, i'm from morocco as well, i need your support to install asterisk server with A2billing and gsm dongles, are u intrested ?
Hi cvierilviv, I noticed your profile and would like to offer you my project. We can discuss any details over chat, i need you install for asterisk and free pbx or other solution for usb dongles for voip termination could u work on it and will need support as well after the project so what do you think ?
i need dongle expert who has knowledge in asterisk configuration, with dongle looking to solve an issue we are facing. The required should have already worked on configuring dongle with asterisk. Below is the log from asterisk CLI we are facing. This is first 3G modem we are trying to connect, we need to connect 3 more, in total 4 modems. WARNING[30258]: at_response.c:384 at_response_error: [card0] Error checking subscriber phone number -- [card0] Dongle needs to be reinitialized. The SIM card is not ready yet -- [card0] Error initializing Dongle WARNING[30259]: at_response.c:384 at_response_error: [card0] Error checking subscriber phone number
I have a setup where I have two asterisk connected with each other on openvpn iax trunk, one asterisk has public ip and other has a private ip ,the trunk is working fine ,only problem is that call quality is very bad ,I need some one to help me fix the voice quality ,my budget is 30$ for this project ,please do not bid high amounts I will not reply to them .Also please bid if you have worked on such scenario and you understand asterisk.
We have an asterisk 4 voip server and during 12.30pm - 1.30pm our phone system goes unanswered due to everyone being away on lunch. We need to following coding in: 1) A key combination pressed on the phone e.g #6 dial that meaning that the called now receives a message (we will get this professionally recorded) saying we are receiving a high volume of calls and the caller can press a number to leave a message which we receive by email in wav format. 2) another key combination that enables night mode so at 5pm we can dial e.g #8 and when called they hear another recording saying our offices are now closed our opening hours are . . . etc. We will also have this professionally recorded and need a voicemail option we can again receive via email
Software for a call center to make invoices for clients and credit slips for agents. We use FreePBX - Asterisk for the PBX. The software can be developed or open source.
Good morning, the job consists in installing asterisk and vicidial on the server to be able to use them for a call center. Predictive must also be activated.
Hello, We looking to install and configure Asterisk and SIP for calling facility for auto dialer and predictive dialer. we are using Vtiger crm and i have already installed vtiger crm you need to install the other all required and configure this for us. or we can use gotodialer also, or you can suggest best that can give us quality call. Please BID only if you have good experience and knowledge in this. Thanks!
How can Fix FAS Problem With CDMA Termination using Asterisk
I am sending a call out using a call file after changing the caller id using Local Channel. The call is then connected between two parties. After talk is over the channels are not freeing up from core show channels . I need assistance in freeing these channels so that hangup script can be called and I can do billing.
Asterisk and FreePBX with WebRTC for Inbound call center. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk w. WebRTC, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. See attachment for issues to be solved. The system must be checked to get it ready for production. The physical Servers (3) are installed and most of the software is also setup. A late Ubuntu OS and Teamviewer is required to login to our servers. --- Project will be divided into milestones, to be discussed.
Hello, I am looking to setup sales call center buy using some opensource like Vtiger crm Sugar crm Asterisk PBX VICI dialer. I need expert freelancer who have good experience who can setup this all for me. So only expert experience freelancer needed for this project who have experience in the above my said list. In your suggesion you need to explain me which one we need to use and why? I have installed Vtiget crm already. Thanks!
Integrating plivo with freebpx / asterisk to build switchboard support simbox and hotline
want to make IVR route asterisk based need following features I need Asterisk Working as voip gateway. 1) when i send call on my asterisk server after few Rings ( witch will be random some 3 rings some times more or less) call pick up and start random IVR ( Need to upload at least 30 to 70 Different Voice Recordings). 2) when i send calls on Asterisk server put on conference e.g ( i send 2 calls both will connect with each other and if i send 4 calls 2 / 2 calls connect on conference & if 1 call drop and 1 is connected then connect next call with old call) CONFERENCE CALL 3) When i send call on Asterisk server there is must be Voicemail e.g Caller call on number 123456 call go to asterisk after some ring there will be IVR like that " 12...
I need help configuring my JeraSoft. I need someone to help me from start to end have a successful JeraSoft Billing with my Asterisk using Teamviewer to teach me how to make JeraSoft work.
want to make IVR route asterisk based need following features I need Asterisk Working as voip gateway. 1) when i send call on my asterisk server after few Rings ( witch will be random some 3 rings some times more or less) call pick up and start random IVR ( Need to upload at least 30 to 70 Different Voice Recordings). 2) when i send calls on Asterisk server put on conference e.g ( i send 2 calls both will connect with each other and if i send 4 calls 2 / 2 calls connect on conference & if 1 call drop and 1 is connected then connect next call with old call) CONFERENCE CALL 3) When i send call on Asterisk server there is must be Voicemail e.g Caller call on number 123456 call go to asterisk after some ring there will be IVR like that " 12...
I need a mobile app that will be used as an extension / soft phone, I mainly use Asterisk & FusionPBX and need it to work with at least these two. Would need it to be reliable, easy to configure and branded. For both iOS & Android.
Hi Ambiorix, I want to discuss with you a web application cloud based call center website to connect local telecoms then on-board clients, is this something you can do using asterisk and vicidial?
Asterisk-FreePBX with WebRTC for Inbound call center. This is only for freelancers with a very broad knowledge-base and who know how to get things done on our servers. First things to fix are, attended transfers and PBX calls agent that did not register for his shift. Some requirements: FreePBX/Asterisk w. WebRTC, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. There are a few more fixes to make and the system must be checked to get it ready for production. The physical Servers are installed and most of the software is also setup. A late Ubuntu OS and Teamviewer is required to login to our servers. --- Project will be divided into milestones, to be discussed.
