VoIP Jobs

we want to write a script which can check to see if a phone number is valid or out of use

We need an asterisk expert to analyze our CDRs, asterisk logs, queue configs, and call recordings to determine why 4 calls on Monday skipped an available agent in one queue (didn't even ring them) and moved on to the next queue which is an undesired behavior.

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 [login to view URL]

1: Functional characteristics - Individual RTP media streams shall use an associated RTCP channel to report transmission performance. - The media transported in an individual RTP media stream shall be as defined in the associated SDP description carried over the SIP signalling link. - The SIP signalling shall carry embedded ISUP signalling in accordance with the ITU-T SIP-I specification Q.1912.5....

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...

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 ([login to view URL]) 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 ...

Hi. We use ASTPP ([login to view URL]) 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 ...

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....

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 [login to view URL] 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.

I need a freelance Developer/ Programmer for VICIdial. Must have 100% knowledge on VICIdial platform and IVR based calling. Rate per hour is negotiable. Working hours may excide 15 hour.

if you have Android Voip app source code, please bid. I will discuss details . Thanks.

I need to add call history and contacts function.

We are looking for someone that can assist us with setting up Vicidial for Outbound Calling Agents.

Hello i need FreePBX + Asterisk + WebRTC websocket config on linux server.

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

Ich habe derzeit unter einem Server das HYLAFAX installiert und dazu [login to view URL] 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: [login ...