Freeswitch voicexml asterisktrabajos
in my asterisk moh files are played but cannot hear anything using g729 freelancer to help resolve this issue.M budget is 20$ for this ,please do not bid high i will not entertain such bids.
We need an API, which can be called from asterisk (custom configuration) that would let us integrate it with zendesk so that it provides functions similar to: /apps/five9-for-zendesk/ /apps/asterisk-dtk-by-cdc/ The module should meet the following minimum criteria: - Must be installable in zendesk as an app - Must be able to assign zendesk agents to extensions - When an incoming call is coming in a popup should appear in Zendesk containing the following 2 scenarios: 1) if the customer is in zendesk the customer name should appear with an open profile link and reject call link, . 2) If the customer is not in zendesk the phone number should appear with a create contact link. - If the call is answered a new ticket should be made with relevant info already populated. - If the c...
Seeking Java/JavaScript/HTML/Asterisk developer(s) to make a web-hosted VoIP API endpoint. Overview: We want to make a VoIP solution for an open world game. Players on our servers have an x, y, z coordinate and a room r. We want players to also have the option to open a web browser, navigate to a URL, and join their voice room (r) with other players. There will be multiple voice rooms. Those close to each other using x, y, z coordinates will be able to hear each other, but the volume of each fades over distance. This system should be pretty abstract. At minimum, we should be able to make calls to a Java web API (maybe running Tomcat) and be able to update player's x, y, z, and r values. What we need: 1. A self-hosted VoIP instance of Asterisk capable of communicating ...
need to install and configure a system that will allow termination of sms using 50 modems connected to a usb, actually i use linux with chan dongle and astersik fro voice termination, but it does not support sms termination. if it can support it it will be great. otherwise i can use a different module/platform, like freeswitch, Ozekisms, or also Diafaan configuration on windows is acceptable.
Looking for an asterisk / freepbx developer who has experience in integration freepbx and IDS PMS
I have a Cisco phone model DX650 and i am trying to make it work on FreePBX Asterisk. I need an EXPERT in Asterisk and in CISCO IP PHONES who has Already Done this before with success.
Hello I have on AWS instance a freePBX server installed, and sip trunk is twilio. I installed Zoiper on my android phone. I am willing to install other apps. If you need to login to server I prefer you use TeamViewer. :) I'd like to have somebody(YOU) to help me configure a softphone app on my cellphone to start making phone calls from it. Please reply with a timeframe this task would take you
I have been running into a weird problem with Issabel. It may not be a problem but I just want to double check with somebody who has better knowledge than mine. Its just an hour max. So, I qoute $10 for this. Thanks
I am looking certified Asterisk/FreePBX tech person who has good knowledge in networking, vpn/vds/Linux. with good skills of planning and be able to start work immediately!
I am looking certified Asterisk/FreePBX tech person who has good knowledge in networking, vpn/vds/Linux. with good skills of planning and be able to start work immediately!
SBC is active and operational all sip will be done on inside, no networking will be done by asterisk engineer sip trunks in and out Auto Attendant set up and recording handsets and users set up voicemail set up explanation of hunt group setup for round robin and collective overall complete system setup of Asterisk Issabel, remote technician available for datacenter and onsite for login in phones. Isabel already installed and tested with one handset. all handsets grand stream
we need a developer that have the capacity to create new CHAN Module that will work with new voip Hardware
need expert to insatll g729 codec for pass calls.
Hello, need voip expert , need softswitch with 5000 CC , no reporting , no extra feature . One Customer >> My Switch >> ONe Vendor
VoIP developer requires good knowledge of voip development free switch and asterisk developer .
Hi Web&Mobile App Developers, my name is David Ortega, I act in behalf of a spanish voip company. We are looking for two developers with experience in react, node, freeswitch and docker. Experience with Elasticsearch would be nice also. It's a long term position. Pur budget is about 500$/month each. Please ping me if interested! Thanks a lot for you time.
Hi Saif, my name is David Ortega, I act in behalf of a spanish voip company. We are looking for a developer with experience in react, node, freeswitch and docker. Experience with Elasticsearch would be nice also. It's a long term position. Please ping me if interested! Thanks a lot for you time.
Hi Arpit, my name is David Ortega, I act in behalf of a spanish voip company. We are looking for a developer with experience in react, node, freeswitch and docker. Experience with Elasticsearch would be nice also. It's a long term position. Please ping me if interested! Thanks a lot for you time.
I am looking for an EXPERT with opensips and astpp with multiple freeswitch to set up a complete switch in a cluster with centralize DB and billing or . When optimize the only limitation of Concurrent calls and cps would be hardware You must use the latest stable versions of everything. you must provide sample of similar work to be considered for this project. Do not waste my time if you have not done this before
I am the co-founder of a medical tech startup. As part of our HIPAA compliance readiness we need to create a disaster recovery and business continuity plan. This project is for the creation of this document, preferably with minimal involvement for us aside from answering questi...and the contingency plans we have. I prefer to use Google Docs for collaboration. The way this works ideally is you send me a list of questions you need answers to in order to fill in the non-boilerplate aspects of the DR plan and I answer them for you asynchronously. When bidding for this project please link to a similar document you have previously created. Please also include the keyword "asterisk" in your bid comments to demonstrate you have read this project description in its entirety. T...
We have setup ASTERISK working fine incase of callback, we want to connect agent first on call(agent should get ringing ) then will connect and then call with customer.
Hi, PLEASE READ CAREFULLY THE WHOLE OF THIS BE...THE WHOLE OF THIS BEFORE BIDDING We have a project for an Asterisk IP Pabx expert to configure a hosted phone system in the cloud. We require up to 10,000 extensions eventually. These extensions are to provide emergency dial out facilities for an emergency call system for elderly people. Each elderly persons home will have a unit that looks like an IP extension off the PABX. When an emergency call is made (by pressing a Help button), the IP Pabx will make a call and allow the relative or help desk to talk to the person who needs help. We want to initially set up an Asterisk system to test the service, then develop a more sophisticated service. YOU MUST HAVE ASTERISK EXPERIENCE Please make the title of your bid ...
Dear Freelance, We are looking for Asterisk and Laravel engineer to help us build a system of billing. This system will provide billing to end user. It is a customized setup and it will retrieve data from AsteriskCDR, A2billing database and FREEPBX database to create a end user billing platform.
Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File:...incoming calls, 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: PHP, Software Architecture, Asterisk PBX, MySQL, JavaScript See more: making money with international premium rate numbers, iprn telecom, iprn providers, international premium rate numbers providers, international premiu...
Hello, We have a working FreePBX distro installation working since 2 years. Now we want to do following enhancement in our FreePBX server. 1. We want to use our FreePBX server for internal office text message communication. I know this can be possible using XMPP module of FreePBX. But don't know, how to configure it. You have to do that. On the other hand we want to transfer file as well at the XMPP messaging window. I don't know whether it is possible or not using XMPP. 2. We want to do video calls between extensions. Need to do video conference as well. You have to configure our FreePBX to do so. 3. We are using Bria Professional as SIP client. You have to configure Bria with FreePBX so text messages and video calls could be done as well. Regards, H. Sattar
Hi Harmohit S., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I want to setup click2call on our asterisk based Vitalpbx.
Hi Team, I need someone to setup click2call on our asterisk baes vitalpbx.
We have two fresh installations of vicidial and goautodial. We would like to configure one to test some outbound voice blast campaigns. We would like recommendations on which product to use for our needs. So Goal is: 1- initial trunk setup to asterisk or sip trunk setup. 2- Setup 2-3 campaigns based in one file txt each with all non agent "virtual/voice blast" campaigns. We have 5 different voice messages per Campaign messages which will be based on a specific value on the list. We have text files with caller information which we can convert to csv format. We would like to have the process of uploading daily lists automatically (via cron or other interface) from a directory where daily files are transferred/deposited during the night. Also have the campaigns start automat...
HI would like to add an external IP when calls sent from voip switch to asterisk and add and an external ip which would be media ip and sent back the calls to Voip Minutes.
Our asterisk Gsm Gateway software use PCI board gsm modules its Hardware , the software is working fine ...we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base
Our asterisk Gsm Gateway software use PCI board gsm modules its Hardware , the software is working fine ...we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base
Need to tune configuration for iternal domain calls .
Hie I need someone to deploy witch on my AWS server.I need someone who can configure an Outbound call center based to my requirement
I have a grandstream model UCM 65 10 that reboots and connects the voip trunks very well. however after some minutes, the registration refresh is not working properly. need help: this is the error message in the asterisk logs WARNING[1923] chan_sip.c: Timeout on 896482223-292013357-1450912169 on non-critical invite transaction.
We need to setup a ITSP phone number to BigBlueButton conference bridge, the purpose of this is users can join webconference meeting from their Mobile phone(call). For this we need to configure FreeSWITCH to receive incoming calls via session initiation protocol (SIP) from our nexmo(ITSP) provider. We are looking for someone who has deep knowledge in freeSwitch and BigBlueButton.
Hi I need an experienced developer with wallboard design , that understands a call center operation and wallboard requirements also reports , logon and logoff features etc. Thasnk
We have Installed the latest version of issabel with asterisk 16. We Also installed Vtiger CRM 7.1.0 and the connector works succesfully. At the moment click'n'call works perfectly and it's possible to pair an extension with a single vtiger profile. We are also able to see a pop up each time we call a number from vtiger interface, This pop-up only says that the extension received the call. We Would like to have the "incoming call" pop-up in order to see immediately who is the caller or in order to open a new ticket directly from the pop up.
I have odoo v11 community edition. I am looking for a freelancer who can help me to implement VOIP calls form Odoo. I would like to implement function of click to call in contacts.... I dont have any asterix, freePBX or cloud calls system yet, I do have SIP Trunk lines. I have seen many projects that had integrated odoo with FreePBX system, and I would like to do the same. So in short you will be connecting odoo with our freepbx system. the odoo crm, contact, leads, customers and more, should have click to call function, popup for outgoing call. If you know odoo 11 and you already develop/integrate phone system, please let me know. I need this job to be done ASAP. Thanks
I need expert guys who have good knowlege of opensip and Asterisk
I am looking for an EXPERT with opensips to build me a carrier grade platform for wholesale and retail. I can work with cdr without rates for now. Platform must be able to handle thousands of concurrent calls and also doing rtp proxy. Platform must have a did management feature and the ability to provisioning and turn up customers immediately, retail with1 number or wholessle with thousands of numbers. Cdr must be very detailed with sip trace available with the click of a mouse. Basically all the features of the latest version of opensips EXPERTS ONLY. If you have a customized wholesale version of a2b I am willing to look at it also. Maybe doing signaling only with a opensips proxy in front of it
Hi Fellow VoIP experts, We are working on a leading web-conferencing open source platform for online learning. This platform uses Freeswitch with Opus codec through WebRTC; so directly through the browser. We made some tests and the audio quality is a lot better on Jitsi, we want to achieve the same quality through our platform. If you are a certified VoIP engineer or have a lot of experience configuring Freeswitch for web usage with opus codec through WebRTC we would be glad if you could help us. So your mission is to help us understand Freeswitch mod_conf opus codec and achieve a better audio quality result.
Minor mod_xml_curl to fetch dialplan and gateway config for FreeSWITCH servers. Implement Fail2Ban and UFW. Have the APIs in place. Just need some help on how to fetch them and implement them in FS.
Hello, I'm running VitalPBX. Using PJSIP extensions with SIP trunks. All with TLS signalling and not encrypted RTP. When placing multiple calls at once, voice quality is horrible and choppy. I need someone who can look into Asterisk settings for me and possibly do a network capture and understand what is going on.
Hi Seby Francis K., I noticed your profile and we would like to have some help in freeswitch configuration. Would you be interested?
Freeswitch installation need to be configured for round robin distribution
i need a sysadmin to Install A2Billing + Asterisk + FreePBX
Hello i want an expert to install FreePBX + Asterisk and A2Billing.
I want to create a Voip trading platform (like for example: - See Video ) for VoIP sellers and buyers. The system has to work with Kamailio and Asterisk.
Need someone to configure click2dial to make calls. Calls using softphone and freepbx - done. Odoo is connected to freepbx - done. If you know odoo 11 and you already develop/integrate asterisk phone system, please let me know. I need this job to be done ASAP. Thanks
