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    9,383 integrate sip softphone asp jobs fundet, i prisklassen EUR

    ...Android APP that works as a SIP Intercom... A big botton on the main screen the show (CLIC TO CALL ) when clic it will make a VOIP (SIP) call to an espesifi extencion, also if I call the extension it needs to autoanswer. on the config menu I need to be Avaliable to change this values Sip server IP: Sip Port: Sip User name: Sip Password: Number to c...

    €106 (Avg Bid)
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    3 bud

    I need a C...multi-tenant like Bicom PBXware where I can create virtual pbx to resell to my customers. My customers can access his accounts and manage IVR (with easy gui), configure trunks, sip accounts, registrations, hear registrations & billing and so on... Bicom PBXware software Link : [log ind for at se URL]

    €752 (Avg Bid)
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    19 bud
    webrtc Softphone Udløbet left

    Hello, we want create a webrtc audio softphone in a web page connected to asterisk serveur to make calls on landline numbers.

    €584 (Avg Bid)
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    16 bud

    Sip tool that will send at least 1000 calls.

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    4 bud

    Use a raspberry PI 3 as a PBX/IVR 1) PBX system (FreePBX?) with SIP for google voice (& callcentric) 2) 1-3 extension to go to a rj11 jack (preferably 3 if possible) 3) 1 extension to send a signal from the pi to hardware specified by the dev to close a circut, only after entering code (for example when selecting extension 9, it should prompt for a

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    Looking for someone to build me a flooding tool (for testing purposes only). The tool (software, app, script, etc...) needs to flood any United States number with at least 100 calls per minute. (500-1000 per min will be better). Every call must come from a different number (different caller id) either from a list of numbers I will upload or preferred is I will input the area code to the tool (for ...

    €2703 (Avg Bid)
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    33 bud

    I find the softphone is hangup after 32 seconds automatically from the Vicibox server. What is causing the issue?

    €71 (Avg Bid)
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    3 bud

    Add to feature Proxy and Ath. Name

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    1 bud

    Would like to have a design for a on-boarding platform. Similar to the following link. [log ind for at se URL] Looking to out do this process.

    €416 (Avg Bid)
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    61 bud

    Hello Dear i have Opportunity to work together. now i have local FreePBX system using SIP line with 5 trank and 20 extension from our local telecom company STC. And i use this system for CallCenter serivce by Zoiper application, and i have agent work on it in my office. so I want connect the local FreePBX system to anther online system to we can allowd

    €88 (Avg Bid)
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    1 bud
    Cold Calling Udløbet left

    I am looking for an outbound sales team that is capable of each making 100 calls per day. Phone service/SIP Server will be provided. There’s base pay plus commission opportunities. You will be soliciting local businesses of our telephone services. It is required to speak clear English, but this job is definitely not limited to the United States

    €5 / hr (Avg Bid)
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    5 bud

    Add sip input tab to frontend to phone dialer.

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    3 bud

    Dear Amrit M 1-i have FreePBX system on -Virtual machine CentsOS 5.10 . 2- you can login via teamViwer on widows7 that the virtual machine on it and then you can login t...the virtual machine on it and then you can login to cents vis SSH or via VMware 3- there is web controal also for OS and FreePBX 4- we just need to add new setting for new SIP line

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    Hi Sunil I saw that you have experience with SIP and VoIP, on android and iOS I need someone to build iOS and android App. If you are interested we can continue talking on chat See you

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    We have a few Skype Connect profiles with inbound numbers at...script should (persistent) connect to Skype servers, login and answer the calls by returning a recording (depending on the inbound number). That's it folks:-) It's all standard SIP, pretty straight forward. --> $100 FIXED (it's max 3 hours work, MAX, 30 minutes if you've done it before)

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    7 bud

    Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

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    hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [log ind for at se URL] i want to run this program on a router Router Link [log ind for at se URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580

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    5 bud

    Help programming sip endpoints, applications voice & sms in [log ind for at se URL] I need to connect my [log ind for at se URL] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [log ind for at se URL] Prior experience

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    5 bud

    I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

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    7 bud

    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this

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    24 bud

    i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

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    6 bud

    I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

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    5 bud

    i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

    €170 (Avg Bid)
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    6 bud

    you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

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    7 bud

    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

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    Trophy icon Création logo Startup Udløbet left

    ...companies. It's a startup centred on cloud technologies. (Voip, Sip) The name is " Caraïbes VOIP " The main color must be blue like the sky. Je démarre une startup dans la téléphonie sur IP à destinations des entreprises. Le projet est axé sur les nouvelles technologies CLOUD. (Voip, Sip) Le nom du projet est "Ca...

    €49 (Avg Bid)
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    67 indlæg

    ...com/click-to-call/ etc You don’t even need a fax machine…,Receive Fax from customer as PDF in your Email. The system will support setting up/administrate the call center SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability to activate/deactivate service

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    6 bud

    Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable

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    5 bud

    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

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    3 bud

    I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

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    3 bud

    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

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    8 bud

    ...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me

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    I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.

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    WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network

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    Hi, I have an installation of FS/FusionPB...be up all the time -fix problem with ZRTP - it's of utmost importance that ZRTP will work on mobile data as well as WiFi -calls will always be performed between two same softphone apps, so there will be no compatibility issue -any suggestions are welcome and additional milestones will be created if necessary

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    VOIP ASSISTANCE Udløbet left

    VOIP ASSISTANCE SIP DEVELOPPEMENT

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    6 bud
    Trophy icon JT&T Brand Udløbet left

    I’m looking for a logo for business cards and letterhead I’m looking for some flyers so that I can advertise VoIP and SIP services the Name of the company is JT&T Communications we offer best in class SIP and VoIP. And phone system analysis design and implementation

    €194 (Avg Bid)
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    633 indlæg

    Configuration for softphones in a cisco system

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    0 bud

    1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.

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    ...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [log ind for at se URL] secret GZD397XU Host [log ind for at se URL] port 5060 Register String 390415541985@[log ind for at se URL]:GZD397XU:390415541985@[log ind for at se URL] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [log ind for at se URL] secr...

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    ...similar app to amend for us. Do you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications. Zoiper softphone is a standalone client-side software VOIP phone

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    We need to change the visual identity of Linphone softphone. We will provide the logo, the company name. The outcome of the project would be: IOS softphone version published on our Github repo Android softphone version published on our Github repo Desktop softphone version published on our Github repo All versions above should be ready to at least

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    ...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new

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    Lightweight SIP Client with Dynamic CRM which would display the customer information. CRM Field and information would display dynamically and action should collect and send to server Database. Required Features: SIP Call with transfer/conference/hold/Mute facility SIP messenger. Dynamic CRM-Dynamic field header and value display for each call according

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    2 bud

    I need professional help with setting up duplicate ...• 18 Avaya Phones – Model 1416 • 23-24 extensions Needed • 4 Trunk Cards That I see • 5 Analog Phone Numbers Following Licenses currently on system: • SIP Trunk Channels • SIP Trunk Channels • SIP Trunk Channels • Essential Edition • Essential Edition Additional Voicemail ...

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    4 bud

    as the title indicates, need sip installed on cisco 7960 to work with astersik. total 10 phones

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    Hi. We need to configure asterisk on our ubuntu machine. We need someone high skilled to do the job. We will also need to make it visible over internet with public IP. Also we need to configure some IP phones on it. After all, we need a manual with step by step about how to configure it.

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    I have lan calls working fine, but require someone with Cisco/Asterisk SIP knowledge to help resolve this issue with NAT, no audio not working.

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    I require a voicemail d...csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [log ind for at se URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

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