VoIP Jobs

We are looking for a VOIP expert to help us to figure out away to make unlimited phone calls to the United States and Canada for free. With NO volume restriction. We know its possible if you know of a way. We would like to hire you as a consulting. The quality and reliability must be high. If you know how to do a proxy server over google voice, skype or aws or etc and make free quality and reliabl...

I need to build a cost-effective phone system that can make calls and play recorded messages to people that answer the calls. This is NOT for telemarketing. The receivers of the calls are in multiple countries. The call receivers will most likely be using mobile phones. Phase 1: Define the specifications of the system (including software and hardware needs if any). It is fine to recommend any s...

I need someone to call phone numbers provided in a CRM. I do not need you to do much more than confirm that the person on the phone is the one whose name will be provided. Any dead number needs to be deleted from the CRM, any number that is the persons number needs to be highlighted. Very simple and straightforward, need to have access to VOIP/Internet technology to make calls into the USA. Ple...

Vorrei una modifica a A2billing in modo che mi permetta di vendere consulenze telefoniche. Il cliente deve inserire il suo numero di telefono e acquistare minuti di consulenza tramite paypal. Una volta completato il pagamento, il cliente chiama una numerazione geografica che lo riconosce automaticamente e lo inoltra alla coda dove risponderà un operatore. Una volta terminati i minuti dovr&a...

install codec g729 on fusionpbx

I'm looking for A2billing expert who can modify it to work with a Chat live system. It will function as a CDR to process a chat history sessions. API integration to a Chat Live server.

Total installation with all required features for Blended call handling. Need features like IVR, Call broadcasting etc

Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

I need to integrate a softphone with fusion PBX/freeswitch.

Only Experienced Call Centers with a Proven Record in Transfers in Medicare supplement need to Apply for this fixed payout Campaign. Inexperienced Centers with No idea of what a Call Center is,or No experience in Medicare Supplement warm Transfers Need NOT Apply. This is a 1 Year Annual Agreement selected Call Centers will get from the Client. Client will provide Training,VoIP and Database.

Hi, I need someone who is expert on Asterisk and PHP. I need him to integrate Asterisk with Jerasoft Billing by Radius. Please don't forget to mentioned 2+5 in your bid otherwise you will be reported. Need to start asap. Thanks

we are trying to create a script where we can check the validity of numbers using a dongle/modem where as it will ring the number and check its status

Hey Need to add a new feature ASTPP VOIP BILLING ASTPP use freeswitch pbx, but he generate diaplan. wee need add function pitch shift

Hi, i have developed a OBD voice Broadcasting Application on PHP, and hosted on Centos with Asterisk. my Problem is I am not able to send concurrent calls more than 300. If you can help me out then that would be great i also want to connect PRI lines in this application. please quote the price for this.

Looking for some one who can connect JSSip or SIPml5 to asterisk with incoming and outgoing calls,

Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

You will be responsible for: - selling a SaaS fraud detection solution to mobile advertising companies and app developers - meeting sales targets - database creating from web to Google Docs - work organized, know how to reschedule calls, keep notes - be eager to learn You must have: - at least 1 year of cold calling sales experience - an understandable English voice - basic understand of how inte...

I am Peter from Rwanda/ Africa. I am looking someone who can configure Skinny & SIP on our Cisco call manager express. We currently have E1 service which is active as well. Proposed Features: 1. Incoming and outgoing calls 2. Music on hold 3. Call transfer 4. Call park 4. Voicemail 5. Auto-attendant 6. Billing We have different brands of IP phones like cisco, Unifi VoIP phone and Polycom. T...

We are having problem with Elastix call center. Searching someone to fix the issue. Thank you

We are looking for someone who have developed an IVR system before. You must be able to show a demo to prove that you have the knowledge and expertise. The system be able to: Generating outgoing calls handle incoming calls and transfer calls based on user input Provide voice recording capabilities and replay log with time stamps Most provide APIs to allow third part integrators to interact with t...

Need an outbound IVR calling software as seen here [登录来查看链接]

I have installed OpenAnswer on AWS. Now I need to reinstall it on new Server. Only expert is needed.

Need a developer to help us rework out implementation of Twilio. Must have the following knowledge: Node.js, Python. Deep Understanding of Twilio Voice API. Deep understanding of Automation Functions, such a reminder calls etc. Familiarity of Amazon Web Services. Step functions / State machines is a huge plus Optimizing implementation in order to conserve cost. Our platform initiates an outb...

Pjsip2.8 is tested with "pjsip-app/bin/pjsua-mipsel-unknown-elf" and the following error is generated during the call. Alsa_dev.c ca_thread_func: overrun! Alsa_dev.c pb_thread_func: underrun! Cpu occupancy reached 96%. I can provide ssh to let you log in remotely. Codes is wm8960，Alsa drive is ok, I can play music with madplay Can anyone help me find out why? The operating environment is...

Hi Freelancers, I have a hot project for integrating Vicidial with Salesforce. Basic Scope: 1. From vicidial, when a certain disposition is updated on live call, it will submit it to Salesforce as a lead using the lead API. If the lead already has been sumitted, that it just updates the lead and not submit a new one or on salesforce it will treat it as a dupe lead. 1. From Salesforce, when a le...

Se requiere configurar una central PBX Asterisk, para que funcionen las llamadas entrantes y salientes, ademas de una troncal entre varios países

I wanna run voice campaign so i need a solution from where i am able to run voice campaign.

We have a Cisco AS5400 Universal Gateway which needs to be configured via an E1 ss7 . The Cisco gateway would interface with with TDM E1S and SIP Interface. We need to establish the Voice signalling via ISUP Controller which would be configured in the CISCO AS5400 via SLT interface via IP.

Hi. We use ASTPP ([登录来查看链接]) for billing. We need create a feature for transfer calls (Similar to FusionPBX *2). The ASTPP use xml_mod_curl for create the dialplan, once, yours need use the same engine to create the transfer calls. We already test the apps: bind_meta_app for capture the digits and works fine, but don't transfer to XML will make the transfer call (att_xfer). We need you f...

Hi. We use ASTPP ([登录来查看链接]) for billing. We need create a feature for transfer calls (Similar to FusionPBX *2). The ASTPP use xml_mod_curl for create the dialplan, once, yours need use the same engine to create the transfer calls. We already test the apps: bind_meta_app for capture the digits and works fine, but don't transfer to XML will make the transfer call (att_xfer). We need you f...

Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

We need a Voice for our IVR and voicemail.

We are a company specialized in termination. We have many destinations with several operators to handle , each has his own unique system. To keep track of our business and to be able to make sure everything is working properly, we need an expert in cloud configuration to manage the cloud configuration. We know exactly what settings we want to apply, all you have to do is the proper configurations....

Ich habe derzeit unter einem Server das HYLAFAX installiert und dazu [登录来查看链接] als Anleitung versucht zu installieren, wobei das T38Modem klappt und loggt sich ein in die FritzBox oder SIPGATE, jedoch das HYLAFAX hebt nicht ab... also es ist sicher nur eine Zeile in der Konfiguration die fehlt, aber sonst müsste es laufen... also ich habe mir die Anleitung zu HYLAFAX unter: [登录来查看链接] angese...

Need Asterisk expert, Phones are down with new router trying to debug where the problem is

hi i have a ASTERISK SERVER and it have a SIP Account i have a kamailio5 installed on my local system now you need to register kamailio5 to Asterisk using SIP account when asterisk server send call to kamailio5 ==========kamailio5 have to send call to local gateway IP like [登录来查看链接] if you can make it please bid

Hi there, We are looking at setting up FreePBX in our Melbourne office. Let me explain the technicality first Using Twilio elastic trunk Handsets (mainly grandstream (gxp2170, gxp2140) Vmware pro FreePBX 14 The solution we are looking for: We know our requirements and we need a set solution using the above infrastructure that we can replicate in other offices ourselves after you fix the Me...

Hi. We need new calling software. What its need to do: 1. Call to our phonebooks and play audio file 2. Hear and understand speech and convert it to text (Cloud Speech-to-Text Google API) 3. We need to do workflows and when customer say something, software get trigger from text, we need to play pre recorder audio file 4. Record all data in audio file and text file 5. If customer say something i...

Need to develop SBO VPN to compress VoIP bandwith,

i have USB modem i want to pass call using asterisk chan_dongle i have Raspberry Pi and Raspberry Pi have connect USB modem with SIM card you need to pass calls using this USB modem i have now HUAWEI E303 modem if you need any other model i can arrange also need to dial USSD and get output and entry to TXT file if you can make it please bid

We need to integrate astpp core and fusion core so we can send and receive calls from provider when provider setup only on astpp. We have changed fusionpbx from mysql to postgresql. We need now call side integration and web side integration has been done.

Dear freelancer i want to install a VOIP Engine like opensip sems-server Kamailio you can install any one i just want to pas call 1st install Engine on openwrt router 2nd config SIP to pass calls codec g729 router model : TPLNIK 3420 v5

I have fusionpbx sip server working online. I need help to use cisco phones model 7945 with this pbx.

I need someone help me to configure callto link that in windows registry automatically start a batchfile that was written by me.

i need to send call server to local voip gateway like server in USA VoIP gateway in Bangladesh server to Bangladesh Openwrt Router connected BY VPN and VoIP gateway connected to openwrt router so i send call from server its work fine but i need some one who can help me to create SIP proxy to reduce Bandwidth now per call on SIP G729 Using 30kbps but i want to make it under 10KB we need to remove ...

I am looking for a experienced asterisk professional who can develop a voice sms server application using hardware dinstar VoIP gateway. For details inbox me.

I have a bunch of phones that I need to have auto firmware update when they are factory reset. I have a DHCP 160 option pointing to HTTPS server for provisioning. Firmware is also on that HTTPS server but my phones connect to my SIP server correctly and ignore the firmware update task. I need help getting my phones to force firmware upgrade when they come up and provision.

Hello we need a telephone customer service with greeting and leave a message alternative. We have not decided yet with Plivo or Nexmo which is both at $2 per month. We need someone to setup a PHP script to take inbound calls for our website. Our website is Ubuntu.