I want to hire an android application developer for my start up. I only want an Indian who can able to meet me in [log ind for at se URL] we would discuss about the idea.I don't want any organisation to bid only individual or 2 people are can only be part of us.
I am looking to setup an application or script that will run to sync contacts from googles contacts directory on one of our google accounts to a XML file that is for grandstream phones. Something that can be setup to run once a day or once an hour to sync our contacts to the phones. The Grandstream phones will then sync to this XML file that the script or app updates.
we need the sms and IVR developer for our projects the below details are here: SMS- suppor...send and receive multi SMPP account billing for each transaction log generation filter and rules apply DND (Blocklist Feature) connected with MySql IVR- IVR flow CDR sip account call record billing option the criteria based on portal or application
Hi, We are building a VOIP project which requires SIP Trunking methodology and integrating SIP Trunk with Asterisk server. (Asterisk version is important, while you posting at which version can you able to deploy) - Need to have excellent skill set in Asterisk, PBX, Integration of SIP Trunking - I must able to make calls thru Asterisk. I will give
I want the page to be on Top of Google without Adverts. Linkbuilding, SEO Word setting, Forum postings, traffic leading, Facebook page etc.. Pls write suggestions how you going to make it no. one?
WE WOULD LIKE SOME OF OUR IP PHONES TO BE PROGRAMMED. WE HAVE 15 MITEL 5224 AND SOME ,...OUR IP PHONES TO BE PROGRAMMED. WE HAVE 15 MITEL 5224 AND SOME , CISCO SPA 504G PHONES. THEY NEED TO BE PROGRAMMED. ALL TOGETHER WE HAVE 25 PHONES THAT NEEDS TO PROGRAMMED FOR VOIP. YOU MUST BE FAMILIAR WITH MITEL PHONES, IF NOT THEN PLEASE DON'T BID FOR THE WORK
...Maintain and manage Genesys Routing, Framework and reporting. Responsible for supporting call center routing strategies and have experience with Genesys reporting, URS routing, SIP and GVP technology. Good understanding of Genesys (CTI) infrastructure. Strong working knowledge of IRD and CME. A working knowledge of Genesys Systems Architecture to create
Need install asterisk and sip server on virtualbox for receive calls directly to computer and if 1st line is busy automatically forward to second free line with call recording and without monthly fee
...+91-XXXXXXXXXX(Agent or user) come to my SIP and thus to my dialplan. If Agent calls he should be loged in as an agent If user calls he should be transferred to a Logged in user. Rewording the above requirement. I have my Agents say A1,A2,A3 on remote location. I have my callers C1,C2,C3,C4,C5 too on remote location. I have a SIP installed on my server which
We are using the Asterisk PBX With a Linphone SIP client in a Linux environment operating on the Olimex A20 and PINE64 and are experiencing very high echo. We understand Linphone uses the Speex Echo Canceller. We either do not know how to adjust the echo canceller or we need to substitute it for another excellent echo canceller. We would like someone
I am looking for someone who can customise the UI on Linphone for iOS, Android, Windows and Mac in that order. We will be providing all of the re...who can customise the UI on Linphone for iOS, Android, Windows and Mac in that order. We will be providing all of the required graphics and we also have an internet facing SIP server it will register to.
Hi altr, I'm looking for someone to help me r...using freeswitch for wholesale scenario, When i receive 181 call is being forward sip message from supplier, it is being absorbed by freeswitch and not sent to Leg A. I don't want to spend time to figure it out, i just need you to tell me how i can make sure this SIP message is forwarded to the customer
Find me some Leads for a VoIP Service. The Company offers cheap rates for VoIP calls via Mobile App (Andorid, Symbian, IPhone). Please, I do not need Email Addresses to send bulk Emails. I need some traffic twoards my site. And acctual leads that generating paying customer. Any suggestions appreciated.
Asterisck pbx server installation (fusion or frepbx), configure and integrate with Odoo VoIP module v10 for Web based calling using WebRTC protocol. Need proven or similar project experience in Odoo platform.
I need a software that allows me to make phone calls on the internet from a free VOIP service. You must find the free VOIP service for me. I should be able to change the phone number from which the calls are made to random numbers each time a phone call is made. I prefer the code in VB.net but if you can not do it in VB.net please explain which coding
...script will be provided but you will have to update it with notes. - Qualifications: I need someone that speaks great English, has organization skills, experience with VOIP (SIP) or has a similar program already so that we can check. Duties: Negotiations with the top officials of the companies; Sale of services in the field of B2B; Requirements:
Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload
...only bid with in my budget don't waste your time on chat if you bid less and then increase it after dont bid Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details
Hello, We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows/Android, or by using the
Install VPN and setup a server connection Install OpenSIPS with graphical user interface install codecs connect GSM gat...connection Install OpenSIPS with graphical user interface install codecs connect GSM gateways to server setup OpenSIPS billing module install fail2ban and configure configure sip connections with clients and perform test calls
...back end provider (SIP or otherwise) and server details.(ie. Centos with WHM and cpanel running Asterisk) I already have some hosting options in mind and I prefer Centos with WHM and cpanel, running various services to accomodate the VOIP server and the website. Basically, we need to figure out what voip server and back end sip or trunk providers
Skype Connect has the SIP trunk feature to use Skype as a SIP trunk of PBX. I tried to configure Skype connect with FreePBX but couldn`t make it work. If someone can do that and already configured such setup, I`m ready to pay.
...from iso on a bluehost server. I can register my sip phone, I can make inbound and outbound calls. The only problem is there is one-way audio on the call. If I call from my Windows Sip phone to my cell phone -- My cell can hear me, but I cannot hear the cell. If the cell phone calls my Windows Sip phone -- The cell can hear me, but I cannot hear
I am looking for provider who provide me sip gateway for Indian operator but with open caller id and API. My concern is to send broadcast pre recorded voice.
I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements
Asterisk PBX application to perform the following. To setup a SIP call between two servers using a specific codec. Have one side playback audio files. These servers will be connected over a WANem emulated link to induce packet loss/jitter/delay etc and produce degraded audio. Have the other record the degraded audio and store it .
Build a VoIP dialer app for Android/IOS and WM using either Linphone or Csipsimple to include the following features. - Account auto provision/Signup via REST. - Account Balance Check Via Rest. - Account Topup via rest.(In App Payments) - Multiple Sip registrar options (Selectable) - User Credit display in app. - Voice/Video/Chat support - *** SIP Header
I am looking for Cisco VoIP Engineer who has outstanding knowledge on Cisco tools. Has to help me with projects/training. I am open with pricing part. Please feel free to reach out to me.
Dear all, We have a urgent Vicidial system to se...information and agent disposition data into one outlook. That means initial upload of list information must converge with output of agent disposition to create one report 6) To create sip trunk for testing 7) To create sample lead list 8) Testing and commissioning 9) Agent and supervisor training
SIP dialer for Android and IOS. Should support most Android and IOS versions including IOS version 12. Features: Runs behind NAT Support Bluetooth Echo Cancellation Push notification Call history display Account balance display Address book integration Support UDP and TCP protocols Support users to Login or Signup Supported languages: English, Arabic
1.I need someone to check my lan for see if it’s secure, and set me up firewalls ( pfsense ). [log ind for at se URL] check cisco switch and routers [log ind for at se URL] set up 2 voips phones 4. Set up windows servers
Need an Asterisk/FreePBX guru to show me how to setup xlite and Microsip sip phones to register with FreePBX extension.
...IPPBX. The main goal is to: 1. Setup basic IVR. Only one level. 2. Setup of 2 profiles of users. a. Will be able to make local calls only b. Will be able to make local and external calls 3. Configuration of 2 SIP and DID services 4. Configuration of Conference rooms (phone bridges) 5. Configuration of VOIP VLANat the IPPBX and Phone level (network