Blackberry sip softphonemunkák

We, here on VMAX Digital, want to integrate our Softphone App with Push notification services. And for that, we need to setup a Flexisip push notification server.

We need a research, someone who knows well about VoIP market, provide us with a list of 50 VoIP (SIP Trunking Providers) who are easy to signup online (no need to communicate with them by mail etc.) and also they provide on their interface a full pricing list with all (A-Z) codes rates.

Hi. Do you happen to have a SIP client that runs on Arduino/esp32, both inbound and outbound? Peter

Do you happen to have a SIP client that runs on Arduino, both inbound and outbound, and working?

Embedded mobile device < develop> - relates drivers(WIFI/BT and ALSA, GPIO, LCD) - Application < Important> - It is not an application engineer for the Andoird smartphone. It is an embedded device. - SIP and WiFI/BT etc... wireless export - communicative to make asta UML - If you do not have the experience, don't contact.

We need a Unified Communications software system that can be hosted on our server at our data center that provides Secure Encrypted Communications with the following features - SIP Voice Chat Video Screen Share File Transfer Windows Mac Desktop iOS Android App

...requirements, 1. Kazoo does not have tariff need tariff in this with all type of customers. 2. Kazoo does not have Device provisioning in it for SIP devices as per SIP brands to program certain keys. Modules Required, 1. Retail Client Client for selling DIDs & Termination routing like VOIP Switch,asiign as many extension numbers to one extension,ver cool feature. to select Different trunks for different PBXs to select number of registrations per account. option must be in Retail & PBX. user interface for PBX admin & Client & Retail client. 8. Live view of SIP call with information of error. 9. SMS service like SIP simple 10. SMS delivery over DID both ways, inbound SMS & outbound SMS. Please contact only developers with the ...

I need a lightweight and WORKING version to perform inbound/outbound SIP, preferably uLaw.

we are looking to develop a plugin/ addon for chrome or firefox that can turn phone number on the PC browser into links that once clicked will call the number and ring on user SIP soft phone (bria for iphone) we will also need to be able to type a number and dial it, review current calls, review call log and have notification on PC screen of incoming call including caller id. only looking for qualified developers with proper experience please send sample of similar work you did

I need you to develop some software for me. I would like this software to be developed for Linux using PHP.

...end-to- end project planning and implementation from scoping, to activity sequencing, effort & cost estimation, risk analysis to quality management • Expertise on Contact Centre Solution (Call Centre) implementation • Designing IT/Telecom infrastructure and implementing technology to support large user groups • Comprehensive understanding of networking concepts pertaining to ISDN, TCP/IP, UDP, VOIP, SIP, H.323, Telecommunication • Effective communicator & negotiator with strong analytical, problem solving & organizational abilities • Expert levels of problem solving and analytical Ability to analyse complex, detailed structures • Good leadership skills and has the ability to guide and provide technical direction and supervision for a give...

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 ...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,...

We need a "branded" softphone with a modern design . This softphone could be based an already existing open-source project in order to let you work exclusively on "branding" and "design" of the product. The main features we need are: • sip communication, • SIP SIMPLE protocol(for instant messaging) • Branding and "modern design" • Provisioning through API • Presence status (not mandatory) • Video call (not mandatory) We already tested this two open source solutions which work with our pbx, have all features we listed before and both give open source code Linphone: Microsip:

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...

Looking for a freelancer who can provide me a SIP system for outbound calls. Requirements: control over CallerID, dial through Soft phone. If such system exist, no need to construct one from zero. do not have my own server nor cloud

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...controls Enable Lobby feature Enable Recording to dropbox, Google Drive, One Drive, and local machine Enable Live streaming to youtube Enable Transcription(cc), integrated with Google Speech to text API Integrated with SIP, Telephone dial-out (India number only as of now) IVR integration Dial-in to the conference Integrate video into a react web application via iframe API Build a Chrome extension for Google and O365 Calendar Provide a Plug-In to Outlook containerize all the roles (signaling, Authentication, Media, Recording, and SIP) Customize JVB and Jibri to autoscale via Kubernetes Enable monitoring via Grafana and Prometheus Configure the infrastructure for scale and cost-effective Deploy video bridges across Geo locati...

We did SIP peer trunking between two pbx, Mypbx and Avaya. from Avaya we are able to receive calls but from mypbx we hear " all circuits are busy try later" message. kindly anyone helps me with this to solve.. regards

Hi Amal A., I noticed your profile and would like to offer you my project. We can discuss any details over chat. "Need assistance with configuration of Vicidial dialer settings and visual customisation. The server is already installed and working with carrier configured, sip registration up and calling working via configured phones in vicidial. I require assistance with configuring the dialler settings and further customisation to make the dialler & agent experience similar to the existing cloud provider I am using. Existing cloud provider is accessible through admin panel but no console. New server has admin, console & teamviewer access (Server is virtualised with low load/requirements -manual dialing/~5 agents.)"

Need assistance with configuration of Vicidial dialer settings and visual customisation. The server is already installed and working with carrier configured, sip registration up and calling working via configured phones in vicidial. I require assistance with configuring the dialler settings and further customisation to make the dialler & agent experience similar to the existing cloud provider I am using. Existing cloud provider is accessible through admin panel but no console. New server has admin, console & teamviewer access (Server is virtualised with low load/requirements -manual dialing/~5 agents.)

Hi! I noticed that you were listed as the winning bidder for this project: https://www.freelancer.com/projects/linux/basic-server-side-kamailio-sip/?ngsw-bypass=&w=f Did you ever get that done? I'd be willing to give you $100 for the same script if you already have it. If you didn't wind up doing it, we can discuss what it would cost for you to do the same project for me. Thanks, Dan May

...Videobridge to provide high quality, scalable video conferences Jitsi Videobridge (jvb) - WebRTC compatible server designed to route video streams amongst participants in a conference Jitsi Conference Focus (jicofo) - server-side focus component used in Jitsi Meet conferences that manages media sessions between each of the participants and the videobridge Jitsi Gateway to SIP (jigasi) - server-side application that allows regular SIP clients to join Jitsi Meet conferences Jibri - set of tools for recording and/or streaming a Jitsi Meet conference that works by launching a Chrome instance rendered in a virtual framebuffer and capturing and encoding the output with ffmpeg We need this to be installed, configured, and handed over to our team Please share previous successfu...

We are porting our existing SIP applications from using the Radvision SIP stack to using PJ SIP. Because of this we are looking for an experienced C++ developers that has worked with PJ SIP. The company is based in Phoenix, AZ, U.S.A so we would prefer a developer that is within the U.S. Any applicant must have at least 5 years using C++. The applicant must know SIP and work in-depth with PJ SIP. Phoenixsoft develops a soft-switch that provides, prepaid, Class4, and Class5 services. There are three application that make up with SIP interface. These are our SIP Proxy, SIP Registra, and the SIP Call Processing layers. All the application run on RHEL/CentOS 7 and are multi-threaded. It is estimated thi...

android and IOS app, the user login using a user and password, sip server has to be hardcoded in the app, once he login a "call Us" button should be there when he presses it just call a hardcoded number on that button, no need for the key bad on the main screen only after the call initiated the key bad is needed except log out, our logo will be provided below is a picture for what am thinking about. the APP's should be uploaded to my IOS and android store.

I am looking for someone to assist me to build a web based interface using React for a softphone used to establish voice calls through a browser. The size of this should be responsive. Width is generally fixed, estimated to be about 200px, height is about 400px but may span the length of the entire page. Elements should be centered. It does not have to be functional in the sense that it does not need to be able to actually make the calls. Instead, I am just looking for a frontend interface with some placeholders that will allow for javascript functions to be added in at a later time to perform the necessary functions (i.e. perform the actual call). Essentially, the softphone would look something like the dialer on an iPhone and provides transitions across various different ...

Configuration of Agent Softphone on an Installation of Queuemetrics reporting running on freepbx server Installation of the components already completed and running....The only remaining aspect is configuration of the softphone due to the https aspect for tomcat. We require someone with adequate knowlegde of the subsystems to successfully configure the Softphone on the Agent Wall as well as on the agent page.

We are a new company in the UK building garden office pods. We need an architect to draw up 2D and 3D CAD designs based on hand drawings. The designs will be for SIP board construction and will need to be heavily detailed listing all the materials, size and measurements for the manufacturers and construction. The work will be ongoing for the wining bid. We will need to see example work before any bid is accepted. Design examples and a full set of requirements will be provided.

FreePBX V with Twillio SIP trunk using PJSIP driver. Calls drop after 30s, narrowed down to not receiving an ACK in response to initial OK after INVITE on incoming calls. Require diagnostic of the reason why the ACK is not being received (Twillio PCAP suggests that ACK is not being sent at all) Suspect issue is related to Contact Header IP not being using properly. Will provide remote access to Freepbx Gui, Commandline, and Twillio Console remotely. PCAP will be provided to suitable bidders. Needed ASAP on business critical line.

-Need a system to convert incoming whatsup calls to SIP destinations. -Need a software to make and receive WhatsUP call from laptop (without the use of the phone active).

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.

I am looking for a freelancer able provide a system to do Voice Broadcasting System. It should able integrate with SIP Phone to enable to do call out.

I need a professional who has experience in PFSense to create a configuration for forwarding to internal servers according to the external domain. We value information security and would like access to PFSense to be internal only. We also use SIP channels, we need to pass these channels through PFSense. If possible, we would also like an installation of a network and server monitoring tool, but this is not a priority.

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 am looking for an EXPERT with opensips to build me a carrier grade platform for wholesale and retail. I can work with cdr wit... 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 f...

Hello, I need a FortiGate VOIP Expert to take a look at my firewall. I use a NON FORTINET VOIP SIP server and voice quality is horrible. Thanks.

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.

Configuración siptrunk 3CX proveedor CallmyWay Costa Rica Necesitamos configurar el sip trunk hoy

********please ONLY Adobe Connect Admins and EXPERTS send me message!!!******** we need create SIP phone pods to adobe connect hosted (on promise) server. *******you must show me worked phone connected room before buy services.********

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

I tried many ways to get US, UK , Australian virtual numbers to Oman but I am unable to get any incoming calls .. Need working techniques to get the calls here without using VPN . I already have US google voice number but I am unable to get incoming calls with that as voip is not supported here Need working technique from tech geness

We have a newly installed FreePBX server hosted by Cyberlynk. The PBX was setup by our SIP trunk provider. Currently the UniFi phones receive calls but with no audio. The UniFi phones send audio to the other caller fine. We need someone who has experience with Hosted FreePBX AND UniFi VoIP phones to assist with resolving this audio issue. There's not much documentation on the UniFi phones and they are difficult to work with so please only apply if you have this exact experience and can resolve our issue.

We are looking for a programmer with experience in working with audio/video (A/V) SIP VoIP communications using WebRTC and WebRTC mobile, call managed through a PBX (preferably Asterisks). The first and main engineering task is to establish A/V SIP VoIP communications from a Pine64 SBC running Linux Armbian through a Web Browser to a smartphone (iOS/Android). Asterisks and the WebRTC services are to exist in the Cloud. After the successful completion of the above task, we would also like the to establish a connection from the Pine64 to a windows 10 PC (running the Chrome web browser). The objective of this project is to create a reliable communications path accessible to other developers through API calls at each point. These other developers would develop Web/Node.js, iOS, A...

We need to find a coder to create a cross platform softphone (Like Bria from counterpath: ). This should be done with Electron/nodejs, with installation packages for Mac / Windows and Linux platform. The project involves knowledge of SIP protocol and VOIP concepts.

Hi We have installed vital pbx 3.0 and we are looking at moving all our clients over from kiero operator to vital pbx. I'm looking for someone to move the clients over one by one. I will advise you the call flow setup along with a zip file for each client for IRV's, welco...Tennant in vital pbx Along with configuring other vital pbx settings for clients to get the most out of the platform. We as a business will be the first customer to go live on the new pbx. Once we test the platform for a week then other clients trunks can be added and they can go live We have around 100 extention across all clients With your experiance we will also look at other wholesale sip providers to save the most amount of money and provide redundancy for us. We can discuss either an h...

Hello, do you have experience to work Rockchip ARM-Cortex A35 and Android 9.0. It must work Android UI with 11r and SIP application. So, it is IP-phone.

We have a single Virtual Phone Number that we want to direct to our Linux cloud server, most likely running Asterisk. We only need a single account so that when the Virtual Number is called then will be forwarded likely via SIP to the servers IP. This will then play a single Audio file of our choosing. Perhaps later we might expand it into a full auto attendant, but we only really need a single message then the user will be forced hangup.

...use both: firestore implementation suggested by google codelab or a homemade webrtc signaling server coded in nodejs (we have a signaling server wrote in nodejs already running and usable) Tech informations about protocols (webrtc, sip) will be provided on job accepted. Actually we have a working demo environment based on core and critical functions/protocols needed. This a “step 2” implementation to test it on a real case. Codelab used for our demo environment can be found here: rtc-codelab We have a perfectly working sip/webrtc phone server on our DMZ ready for our porpuses....

Build audio and video conferencing site with webrtc and sip using jitsi. Front and back end with good management system.I am looking honest qualified developer who wants to work long term to combine these 2 sites. and blogtalkradio.com. with video conferencing part of and some features from .there are very simple but i need honest reliable people i can work with. NO PREPAY OR MILESTONES. Milestones are total garbage. 99.99% of a project is 0 to me . So if you are qualified and NOT a crook. Lets talk. Your bid is the final price. I will not pay a penny more.

...to be changed as required. I'm sure if you've read this far you know whats required in order to build this. We want to make use of VoIP/SIP Trunks to send SMS as we were doing previously with another business partner. Each SIP trunk could send approc 12,000 SMS per hour which we want to replicate with this. SIP trunk costs around 20 USD per month per trunk, which you need to source. The SIP trunks must be 'pooled' rather than allocated to each account, this way they are maximised across all clients. Longcodes required for each campaign rather than per account, again you are required to source these also. ideally, we need system we can take over and order SIP trunks to system and longcodes as rrequired. Clients to have a 'sendin...