Zoiper työt
Hi All, I want a Mobile App like zoiper or any other android softphone client. we have already web services api for all process for app to server, like incoming, outgoing and other voip features.
We are just looking for someone experienced with asterisk and freepbx who can help us with a few configurations, basically, these are the current things we need to be done: - Enable IM - Enable Video - Enable JITTERBUFFER in all channels / extensions - Some devices occasionally being unregistered (mostly android / iphone zoiper clinets) Asterisk 13.7.2 FreePBX 12 Centos 6 64 bit. We will probably need ongoing maitenance, so ideally we are looking for someone who could help us with our immediate needs and then be available for ad-hoc support every now and then.
Looking for someone to develop a mobile application similar to PortGo, Bia, Arobits, Zoiper, etc. The prototype is complete and the API exists. We will buy and use an existing SDK, so all that's required is to build the actual GUI elements according to the design & prototype. All graphical elements (in all sizes) will be provided. The development works will be to construct the GUI elements to integrate with the API and the SDK. End product will be in IOS and Android, but if you cannot do both, we can source separately. We will require existing working samples as proof of abilities.
App Like zoiper with my branding (VOIP APP)
I need zoiper sdk for android and iphone.
voip for zoiper needed installing. please know what you are doing
I need you to develop some software for me. I would like this software to be developed for Windows using C or C++.zoiper installed on a windows server using flowroute as the sip.
We are looking for a person who has experience with VOIP and Skype and is able to configure VOIP/Skype gateways and/or create script(s)/application which will communicate with our existing application via REST API. Or the other option is to modify existing node.js application instead of use REST API (access to the application sources via g...REST API. Or the other option is to modify existing node.js application instead of use REST API (access to the application sources via git repository can be provided and also assistance from the application developer). Our current application works on CentOs 6.7 and is build on top of node.js with / ; ffmpeg is used for handling audio streams. Our current VOIP software is Zoiper but we are open to other suggestions and solutions.
...have well polished communication skills to effectively and efficiently sell our client’s brand/product in a persuasive yet artful manner. Must be able to effectively deliver scripted presentations and handle objections. Desirable qualifications Self starter Great work ethic Honest, Responsible and Reliable Computer literate Knowledgeable in Google apps, Skype and VOIP systems such as Zoiper or Xlite Working Tools: Working laptop or computer (MAC or Windows) USB headset that has noise cancellation abilities Fast and stable internet service (No USB devices or hotspot connections) Quiet work environment ...
Development of Mobile SIP Dialer using Zoiper SDK for Android and Apple stores. Previous experience in developing this type of application required.
Project Description: We need a voip mobile dialer like zoiper and other popular dialer design that has about 5 screens. (Dialer - Call log - Contacts - Settings - Account) The app should integrate with our softswitch. The main functions of the app will include: 1. Making & Receiving VoIP Calls Over 3G, WiFi, GPRS 2. Should not kill the battery of the mobile device 3. Address Book Integration with local phone 4. User Credentials Save / Change Username & Password 5. Volume Control 6. Protocol SIP (RFC 3261) 7. Media Support RTP (Real Time Transport Protocol) & RTCP (RTP Control Protocol) 8. Transport Mode TCP, User Datagram Protocol (UDP), TCP to UDP Interworking 9. Codecs G.723, G.729, AMR (Adaptive Multi-Rate Compression), iLBC, G.711, GSM. 10. Networ...
App Like zoiper with my branding (VOIP APP)
We are looking to develop a softphone like Zoiper, () or Zangi () some of the features are listed below. Please apply if you have understanding of the project. Access to the native contact list; - Favorites / Speed Dial; - Call History / Contact management; - Voicemail integration / notification (SIP MWI); - Hold / Resume / Redial; - In call contact picture; - Configurable DTMF (SIP INFO / RFCs 2833/4733 / RTP Inband); - Notification Center Support for missed calls, voicemail and registration status; - Blocked Numbers List. • Codecs: - Included: G.711 (uLaw & ALaw), iLBC; - Purchasable: G.722 (Wideband), iSAC (HD), G.729 Annex-B; - Support for UDP and TCP. • Other: - Call Waiting / 2nd Call / Call Toggle / Call Transfer
We would like you debug and fix our sip trunking setup to work with our providers so that we may use Elastix MT 3.0 to the fullest with multiple organizations and users per organization. ...providers so that we may use Elastix MT 3.0 to the fullest with multiple organizations and users per organization. We are currently able to have our handsets receive registration, and our trunks are showing registration from our sip provider, however we are not able to make or receive phone calls through the end points (Polycom) as well as we are not able to make or receive calls through softphones (Zoiper) We will provide shell access to the server. We will require a screen recording of all changes that are made by our freelancer. We would like to be able to access it through YouTube p...
We want to integrate the possibilities of VoIP based calling to PSTN (landline) networks in our web app. Our need is a browser based softphone, like 'Zoiper' for desktop. Requirements: - input field for phone no. - call/hangup button - status massage box (calling, hungup etc.) - DTMF tone buttons (0-9 # *) I provide credentials for a sip server to handle the calls, including host, username and password. I have tried the one from , but it seems to be limited.
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Hi Sir ! i have A2billing with Asterisk . Few days ago i have restarted server from that day When calls answered no voice B side . Please can you fix me .Our Clients Need DTMF keys use for every call . They use Dailer Eyebeam & Yateclient , Zoiper. we need all codecs compatible with these dailers and change into G729 and G711 for Providers demand. Please if you have time Fix for us this issue . Server is located in CANADA and clients are in Europe so person who installed in a2billing first he told me its due to server location . But i dont beleve due to server location may case of delaying of ring response and some time calls stuck on CLI Asterisk : Channels . If call answer no voice from party B. some time DTMF not work
Hi Sir ! i have A2billing with Asterisk . Few days ago i have restarted server from that day When calls answered no voice B side . Please can you fix me .Our Clients Need DTMF keys use for every call . They use Dailer Eyebeam & Yateclient , Zoiper. we need all codecs compatible with these dailers and change into G729 and G711 for Providers demand. Please if you have time Fix for us this issue . Server is located in CANADA and clients are in Europe so person who installed in a2billing first he told me its due to server location . But i dont beleve due to server location may case of delaying of ring response and some time calls stuck on CLI Asterisk : Channels . If call answer no voice from party B. some time DTMF not work
Dear all, I am seeking an Outbound Proxy server to be installed on one of my servers .. The main purpose...to a noticeable voice quality issues. So I have searched around and found that there is something called Outbound Proxy that would mainly bypass the ISP's VoIP blockage before sending the signals to my Elastix or any SIP server. In Order to choose you as a freelancer, YOU MUST HAVE deployed this already and you can provide me with the IP address of your outbound proxy server and use it on my ZoIPER or XLite client and see that my extension is registered and calls being sent out with no voice quality issues. Good Luck for all freelancers Note: there was a free service called "http://www.voip-info.org/wiki/view/Free+SIP+Outbound+Proxy+Service" but its no ...
...org/display/FOP/Installing+FreePBX+12+on+Ubuntu+Server+14.04+LTS The objective is to setup and secure (iptables) the FreePBX installation. Once done I will give you details for a DID number (incoming trunk) and a SIP provider (outgoing trunk). You would have to setup both trunks and connect them to an extension for demo purposes. I expect all incoming calls to forward to my extension on Zoiper and all outgoing calls from my Zoiper should use the SIP provider. This has to work flawlessly. CDR reporting should be up and running. Call recording needs to be enabled on both, outgoing and incoming calls. I need to be able to play the call recordings from within my browser (chose the appropriate format for this). Mailbox recordings need to be able to be sent out by email t...
I need someone to setup my zoiper phone to run on my new LD carrier VOXOX and give me a copy of settings for US domestic calls and verify settings are working properly on vicidiaelr for a press 1 campaign
I currently have a PBXww account set up and softphones (Bria, Zoiper etc) working on the system. I have purchased a CISCO unified wireless hand set 7921G which I would like to add to the configuration. I have a TP-Link wireless router in my home. I need the phone configuring for SIP (software update or what ever is required there) and connecting to my PBXww network.
HI I need a developer to create and Softphone, similar to zoiper or eyebeam with diference that can only use or domain, Plese send us, example of your works Thanks
Need a SoftPhone application for Windows based on WPF. The strategy is to develop this in small "step-by-step". Features we are asking for is similar as you find in SoftpPhone Zoiper already has developed. We are using asterisk as PBX. The project should be developed in VisualStudio 2013.
Hi, I have setup freepbx/asterisk with my local sip provider in my own server. I can call with Zoiper to a local number and when someone answer the call - no audio on both end and the call is dropped after couple of seconds. This is the asterisk log file: [2015-06-21 18:34:31] VERBOSE[4468][C-00000000] netsock2.c: == Using SIP RTP TOS bits 184 [2015-06-21 18:34:31] VERBOSE[4468][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5 [2015-06-21 18:34:31] VERBOSE[4468][C-00000000] app_dial.c: -- Called SIP/Cellcom/054444444 [2015-06-21 18:34:35] VERBOSE[4468][C-00000000] app_dial.c: -- SIP/Cellcom-00000001 is ringing [2015-06-21 18:34:35] VERBOSE[4468][C-00000000] app_dial.c: -- SIP/Cellcom-00000001 is making progress passing it to SIP/10-00000000 [2015-06-21 18:34:38] VERBOS...
I have registered with a paid VOIP SIP server called X. At home I am running Asterisk on my Ubuntu server called Y. I am using Zoiper Softphone on my Iphone Z. I want to make outgoing calls from Z through X via my server Y. The setup works. But then it stops working and gives 403 Forbidden on my iPhone Zoiper App. Then later it will work again, and stop working again. Need someone to help me fix this problem so the setup just constantly works. The problem is not at the server side X, as I am able to directly connect to X, via another app called YourDialer from my iPhone, working 100% of the time. This however, does not pass through my Asterisk server. Questions? Let me know.
We have Vtiger 6.2 installed and a Elastix server running. Both are accessable by TeamViewer/SSH. We need the configuration so we can see incoming calls popup in vtiger and we can click a number in vtiger to call outbound by asterisk and softphone (zoiper).
We have Vtiger 6.2 installed and a Elastix server running. Both are accessable by Teamviewer and SSH. Cant forward the ports due to limited access to the network/router. We need the configuration so we can see incoming calls popup in vtiger and we can click a number in vtiger to call outbound by asterisk and softphone (zoiper).
Hi Freelancers, looking for softphones software expert (zoiper or eyebeam) with experience in PBX systems so he can Provide me a Toll Free number with SIP profile including a plan of unlimited minutes and unlimited Extensions for a monthly fee and its configuration and network setup. I also have elastix open source Call Center Software(execution file) if you need it for PBX/extension integration center is made up of 7-8 computers in total running on xp or windows7. Lowest bidders will be considered only. Happy Bidding
We are working with Twilio and have no problem for incoming calls for our clients. What we need is to setup an IP PBX so we can connect our clients through a deskphone system like Zoiper to our Twilio account. We´ll probably need a maintenance for this solution once it´s running.
Need to update and modify ZRTP patch to work with the latest version of Asterisk in pass-through mode. The initial source code and documentation for old versions can be found on In result need to get following 1) Modified source code compatible with the 2) Proof of successful test using SIP clients with ZRTP support, such as CSIPSimple, Zoiper, Jitsi and etc. 3) Step-by-step instructions of the installation process
Need to update and modify ZRTP patch to work with the latest version of Asterisk in pass-through mode. The initial source code and documentation for old versions can be found on In result need to get following 1) Modified source code compatible with the 2) Proof of successful test using SIP clients with ZRTP support, such as CSIPSimple, Zoiper, Jitsi and etc. 3) Step-by-step instructions of the installation process
We have Vtiger 6.2 installed and a Elastix server running. Both are accessable by SSH. We need the configuration so we can see incoming calls popup in vtiger and we can click a number in vtiger to call outbound by asterisk and softphone (zoiper).
We have Vtiger 6.2 installed and a Elastix server running. Both are accessable by SSH. We need the configuration so we can see incoming calls popup in vtiger and we can click a number in vtiger to call outbound by asterisk and softphone (zoiper).
We have Vtiger 6.2 installed and a Elastix server running. Both are accessable by SSH. We need the configuration so we can see incoming calls popup in vtiger and we can click a number in vtiger to call outbound by asterisk and softphone (zoiper).
Looking for Somebody who can help configure ICTBroadcast (based on Asterisk PBX), Zoiper soft phone, Integrate Sugar CRM and do the proper training over a screen Sharing program.
Crear softphone similar al Zoiper pero que este preconfigurado y que el usuario solomanete tenga que ingresar su usuario y password
Development of Mobile SIP Dialer/Softphone (like Zoiper, CSipSimple, 3CX, FRing, etc.) for both Android and iOS. All popular features on the app. Must be able to easily customize some features for resale. Add app to app stores. Previous experience in developing this type of application required.
We would like a simple SIP client similar to XLite / Zoiper. whenever a call is received/is ringing it Opens a URL to a webpage. This can be done using Libraries such as Konnetic.
Wir haben einen lokalen PBX (Asterisk mit Elastix) im Netzwerk, der über eine IP erreichbar ist. Auf XOS und Windows benutzen wir Zoiper und x-lite um mit dem PBX zu kommunizieren. Leider gibt es für Chrome OS noch keinen SIP client. Diesen müssen wir entwickeln. Zu beginn reicht eine einfach Erweiterung, in die ich die Telefonnummer eingebe und die dann den Anruf macht. Später kann man die Funktionen erweitern. Ich weiß, dass WebRTC, HTML5 und JSSIP zur Verfügung stehen, habe aber keine Erfahrung damit. So wie ich es verstanden habe, hängt es auch stark von unserem Asterisk ab, ob es funktionieren wird. Deshalb sollten wir vorab vielleicht ein paar Tests machen.
Necesitamos desarrollar o personalizar un cliente VoIP para Linux (CentoOS) basado en el protocolo SIP. Algunos clientes sobre los que se podría trabajar son Zoiper o Limphone pero estamos abiertos a estudiar otra propuesta. El plazo para la versión beta son 7 días
Design a sip Android application based on csipsimple or zoiper sdk and be able to add gps coordinate into a geolocation header of an invite message when any 3 digits number is dialed according to RFC-6442 location conveyance over sip /presence with ability to share location coordinate with a number just like WhatsApp to other users
Hello, We need to design special Online Support like zoiper or Olark for our web site please tell us the price;
Customizacion de una aplicacion android existente. La aplicacion ya esta desarrollada, solo hay que personalizarla para usar en nuestra empresa. Es una aplicacion del tipo SIP Client para Android del tipo Bria, Zoiper, Sipdroid, etc. Con un tiempo de trabajo de 30 horas aproximadamente.
Go to the following link I want all the features same as Zoiper which would contain voice, video, sip and messages . I will provide you zoiper SDK and you can build the app from it. Your task will be easy this way, I will provide you design too. Thanks
...click-to-call, parallel ringing, recording, MOH, comunicaciones unificadas, billing, ect) – imprescindible. - Instalación, configuración, personalización, administración, resolución de problemas y soporte técnico (Asterisk, Go-autodial, Free-PBX, Asterisk-now, Elastix y A2billing) – imprescindible. - Voip (Códigos audio Gateway, proveedores de servicio, teléfonos IP (Grand Stream, Digium, Polycom, Xlite, Zoiper, 3CX), tarjeta TDM (E1, T1, FXO, FXS) & 3G Hawaii GSM Dongle (E-172, E-1553) configuración e integración. - C, Core PHP – imprescindible. - Asterisk dial plan, Asterisk-AGI, Asterisk-AMI, base de datos Mysql, Windows, Linux (Centos, Ubuntu) - conocimientos. - LAN, WAN, Open VPN...
We have Vtiger 6.1 installed and a Elastix server running. Both are accessable by SSH. We need the configuration so we can see incoming calls popup in vtiger and we can click a number in vtiger to call outbound by asterisk and softphone (zoiper).
...otros momentos la llamada se escucha perfectamente. La troncal esta configurada, las llamadas salen pero , A VECES, no se escuchan bien. Hay que colgar y volver a llamar y al hacerlo , se escucha bien y a veces no. Es decir, el problema es que se establecen las llamadas pero , a veces, no se escucha bien. Las llamadas se realizan desde republica dominicana conectandonos a traves de softphones zoiper a la centralita Voip Elastix en Barcelona (España). El problema tendria que estar en la centralita en España. Llamamos sin problema por voip desde la oficina en republica dominican (via skype o conectandonos a una empresa de telefonia ip directamente). Por tanto creo que esta relacionado con la configuracion del portech o algun problema con este equipò Si sirve ...
...required the server to restored to an earlier date causing Asterisk/Vicidial to not be carried over to the restored database. I am seeking someone who can access my server and get these applications running again. I'll also need a reverse proxy set up so the url to the dialer is not an IP addresses shown to the public. Will also need reverse proxy set up for an already installed version of vTiger. Zoiper needs to be installed. Custom coding needed: a) Code written to let ViciDial upload certain records to vTiger automatically upon successful outcome by agent, b) Code written to output error logs should system crash for any reason Looking for someone quick and efficient with installation. I would like this completed this now if possible. Also- I am seeking ...
We want to hire a freelancer to finish the development of current voip client APP: what current APP already done: 1...what current APP already done: 1. Sip account creation and registration. 2. Incoming call works 3. Chat with supporter 4. SMS verification 5. Forwarding call feature on screen page What need to do : 1. Start with current source code , integrate exist call forwarding, Chat and incoming call to continue this project to final version. 2. Outgoing call (like zoiper outgoing call screen) 3. Contact screen window like zoiper 4. Lanuage (Chinese, English, ) Option,can be expand later 7. Payment (user can buy credit through the following payment) support google wallet but can be extend to others a) Stripe: refer to developer manual b) Paypal