Iax asterisk jobs
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.
i need a asterisk , vicidial , freeswitch , crm expert developer , i need them for long term also to work some hours for day , please bid with you monthly sallary ! thank you !
I want a special setup on our asterisk server. If an agent dials an agreed extension, then the asterisk should automatically dial an agreed number as you can see below. Monday between 8: 00-16: 00 Danish time, then the call should ring between the following two setups. The first time it is dialed, it must dial 004580808080, after 15 seconds, then it must interrupt and then dial up to 004580808081, if no answer is answered again within 15 seconds, it will dial the last number 004580808082. the answer does not interrupt the call so the agent can talk to the customer again. The second time it calls, it must call 004580808081, after 15 seconds, then it must interrupt and then call up to 004580808080, if no answer is answered again within 15 seconds, it will call the last numb...
Hello can you install Asterisk GUI in Ubuntu ? need your assistance to configure and . If you can please message me urgently and ASAP.. Thanks.
Hello can you install ASterisk GUI in Ubuntu ? need your assistance for configuring
Hello, I need your assistance for Asterisk GUI installation.. my server is installed but we need GUI configure..
Hello Vasile, I need your assistance to configure my Asterisk server GUI.
All is installed, asterisk,npm,node js, google cloud you need to integrate speech to text using the v2 key from google
I just need a sip PROXY that will allow me to receive traffic from my customer a USA based telecom company (IP Based authentication) and then send the calls to my vendor who uses an Asterisk and will authenticate my IP. Capacity for 60 simultaneous calls, no rtp relay needed, just signaling handling. Need to be able to download CDRs in csv. file extension. For this project you need to indicate server specifications, I will use services. Also as an additional feature, I will a simple and plain php page showing all active live calls with its current duration.
build the Callcenter via Asterisk server up to 30 extensions welcome prompt Database for call center information IVR and SMS based (give the Details Via chat) technical Integration with Telecuminication Shortcode Routing call transferring / call record / billing record/ DND/ etc few more feature should be mention via chat. looking for an active and responsive guy not only give the details and then he or she busy with other works.
Asterisk 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 with the terminal. 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 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. --- Project will be divided into milestones, to be discussed.
Hello. We need someone who can help us to configure freepbx and asterisk.
Hi Anton, you assisted us with FreePBX setup using AWS singapore last year. I rebuilt the instance in AWS Hongkong - I wasn't able to use the image due to compatibility issue of the network adaptor. Anyway the new instance will freeze (asterisk) every 1 to 2 weeks, and I m not sure why. by any chance you can help us looking into this and resolve the issue? Best, Johnny
Important - no PBX / Asterisk / SIP -> only XMPP CALLING We are looking for a developer who can develop a plugin for Android app to CALL (not push to talk broadcasts) but truly calling via XMPP. When needed also a plugin for Jabber server to enable XMPP CALLS. ONE-ON-ONE / END-TO-END XMPP CALLS between two users as xmpp1@ - xmpp2@ without the use of a Asterisk / PBX server - truly only web based XMPP CALLS. Budget $1000 for the ones who truly have experience in XMPP CALLS.
Hello, We are looking for a developer who can develop a plugin for Android app to CALL (not push to talk broadcasts) but truly calling via XMPP. When needed also a plugin for Jabber server to enable XMPP CALLS. ONE-ON-ONE / END-TO-END XMPP CALLS between two users as xmpp1@ - xmpp2@ without the use of a Asterisk / PBX server - truly only web based XMPP CALLS. Budget $1000 for the ones who truly have experience in XMPP CALLS.
I have extensions which are forwarded to external numbers. I need a method which will allow me to listen to one of these forwarded calls. I will be listening from an inside extension. we'll be listening to a call that came from the outside that was forwarded again back to the outside. I can currently listen to a call before it is forwarded but my current method does not work once the call is forwarded.
Dear all, I am looking for someone who worked closely with FusionPBX in deployment and configuration better yet customization. I have a good experience in Asterisk and Issabel, and for the same of multi-tenant feature which is not available on Issabel, I am intending to move to FusionPBX I am expecting someone who can give me a tutoring session to walk me thru on how to best utilize and familarize myself to use FusionPBX I know that most of the features are in common between Issabel and FusionPBX but the options are scattered over. the best tutoring session would be covering the 5 sections: 1- Home 2- Accounts 3- Dial Plans 4- Applications 5- Advanced on
Dear all, I am looking for someone who worked closely with FusionPBX in deployment and configuration better yet customization. I have a good experience in Asterisk and Issabel, and for the same of multi-tenant feature which is not available on Issabel, I am intending to move to FusionPBX I am expecting someone who can give me a tutoring session to walk me thru on how to best utilize and familarize myself to use FusionPBX I know that most of the features are in common between Issabel and FusionPBX but the options are scattered over. the best tutoring session would be covering the 5 sections: 1- Home 2- Accounts 3- Dial Plans 4- Applications 5- Advanced on
