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    17,932 asterisk sip tls ocs travaux trouvés au tarif de EUR

    Hola con todos Alguien ha podido integar infusionsoft con centrales telefonicas ip como Elastix or Supermicro-Asterisk. La integración debe permitir : - Conocer duración de llamada - Grabar llamada - Origen de la llamada - Numero de llamadas por Gestor y por campaña de llamadas - Consolidación general de las llamadas Para controlar al gestor de

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    Necesito integrar el metodo de pagos de redeban multicolor con woocommerce de wordpress. s necesario que al realizar la implementación hagan uso del protocolo TLS 1.2 a nivel de comunicación. Al igual que si se realizará el consumo del servicio a través de SOAPUI se realicen los siguientes pasos: Buscar donde este el SOAPUI. Buscar el archivo SoapUI-5

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    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    ...nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 1 nr that does not work as i get error sip 503. For security reasons , you will be operating through Teamviewer on my computer which is connected to the remote site of my client where the IP PBX is. I also need to check

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    ...Tabelle (ca. 20 Wörter) Total geht es um ca. 100 Texte in diesem Format zu verschiedenen Hunderassen. Preis also für 100 Texte angeben. Bitte starte dein Angebot mit einem Asterisk (*), um zu zeigen, dass du das Inserat gelesen hast....

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    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

    €46 / hr (Avg Bid)
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    21 offres

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    €729 (Avg Bid)
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    10 offres
    VOIP yate call pass 7 heures left
    VERIFIÉ

    i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

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    app development 5 heures left

    Android sip cellular gateway we are looking for an expert in developing mobile app to develop an app that will expect voice calls using usip server and dial out local number using the mobile network

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    integrate bitrix with asterisk system -

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    ...modify those. Below are more detailed descriptions of the api/ and frontend/ directories. Locations that you will be mainly focussing on as you develop are marked with an asterisk (*). Backend API (api/) api ├── Dockerfile # Description of Docker image ├── [se connecter pour voir l'URL] # NPM dependencies ├── spec # The backend tests

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    Wordpress Plugin for Phone Call Conclusion left
    VERIFIÉ

    Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone

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    ...tools needed to make the box a complete antispam gateway. Installing proper mail server components for multi-domain future use. - Nginx with SSL (let's encrypt) - Postfix with TLS - Redis for caching - Dovecot with Sieve With SSL, nginx, and all the standard tools that we need. Then we should setup a test subdomain so system can fully route and process

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    Need someone to configure Asterisk secure websocket for javascript library integration (such as jssip or SIPml). Asterisk is installed. Access will be provided.

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    Actionscript 3.0 SIP S'est terminé left

    1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

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    PHP SIP client S'est terminé left

    PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done

    PHP
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    ...However, here is the problems that I need your help with to get it to resolved. Problem 1: SMTP Banner Check Reverse DNS does not match SMTP Banner SMTP TLS Warning - Does not support TLS. More Info SMTP Transaction Time 15.598 seconds - Not good! on Transaction Time Problem 2: There are alot of warnings in the Exim Configuration Manager

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    VOIP Project S'est terminé left

    Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

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    Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

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    Local
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    FreePBX support S'est terminé left

    Hello freelancer, I currently need FreePBX installed with our asterisk instance. we have a 5 hardphones, 15 softphone users. We are in need of IVR setup, conferencing, outbound/inbound routes and trunks. Our freepbx will need to be setup based on best security practice and monitored for atleast 15 business days after installation. we are moving from

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    asterisk expert needed S'est terminé left

    Hi, i need assistance of an Asterisk expert on daily basis. I am looking for an experienced expert coder only. 5 star review will be provided for some one Good job! Happy Bidding!

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    Create SNOM Phone Dial Plan S'est terminé left

    I am looking for a SNOM phone guru that can create a dial plan for me with the following...strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765

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    i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [se connecter pour voir l'URL] or yate or besip [se connecter pour voir l'URL]

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    add sip trunk to elastix S'est terminé left

    add sip trunk to elastix i have sip trunk from STC

    €16 / hr (Avg Bid)
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    ...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...

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    ...tools needed to make the box a complete antispam gateway. Installing proper mail server components for multi-domain future use. - Nginx with SSL (let's encrypt) - Postfix with TLS - Redis for caching - Dovecot with Sieve With SSL, nginx, and all the standard tools that we need. Then we should setup a test subdomain so system can fully route and process

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    3 offres

    configuracion de internos y de central asterisk

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    Hi, we need a professional Asterisk / freepbx sysadmin who can fix the nating in dmz using opnsense. currently we have done 90% of work Lan 192.168.X.0/24 WAN A.B.C.D (static public) DMZ 192.168.Y.0/24 PBX is here (also VIP to L.M.N.P public IP) using 1:1 Nating now the odd thing is that we are using linphone as softphone and we are having problem

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    ...documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call T...

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    Template and Invoice Maker S'est terminé left

    To permit clients to fill out forms and documents on our...for the Nginx/Apache Web Server and the SMTP email Server, one for the MySQL Data Base Server and one for the module with LibreOffice. Invoice information comes from another Asterisk Server. To start, Teamviewer must be used, as some software is already [se connecter pour voir l'URL] is a Linux only project !

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    Independent dialer system S'est terminé left

    we wish to have someone connect and configure our [se connecter pour voir l'URL] to our SIP and Trunk

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    I have a trouble with Android 4.4 or less , Android 5 to up is not affected the app it's ok, I require support to fixed by teamviewer , and i have only a couple hour to finish. I use Eclipse, and the trouble is with SSL & KSOAP2.

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    **Must speak both english and spanish** We are developing an inhouse an Asterisk based PBX solution to suit our needs. This is an ongoing project and we will need developers who are experts in PHP, NodeJS, CSS, MySQL, Optimization, Security. Bonus points for mobile development. We also need project managers. Previous experience managing development

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    I have a FreePBX/Asterisk System working at Amazon. I can access it directly or via a VPN. Normal telephony works as expected. 1) Trying to get WebRTC phone (via the UCP) working 2) Trying to integrate external WebRTC Can you help debug?

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    ...capability (CHECK on this) if not we will have to go open source with asterisk pbx, -Company with PBX need to be able to manage DID (phone numbers) and assign number to agents, administrator and super admin account, -Lower the cost per minute and per text (that is why we need to migrate to asterisk open source) -Lastly I need to be able to receive text message

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    SIP面板中找到100家从事房屋建筑的公司 如果事情不明确,请写信给我,我会尽力解释。 在中文里,我通过谷歌翻译写作。 我的母语是俄语。

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    Asterisk & Zoho CRM S'est terminé left

    We have an asterisk PBX integrated with Zoho CRM but it's delivering the channel ID instead of the dialed number to the the CRM extension. We need some one to fix the coding of the asterisk PBX to deliver the correct needed information

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    HTTPS Client - Delphi 5 S'est terminé left

    It must compile on Delphi 5. Must be compatible with TLS V1.0 (If you can make it compatible with TLS V1.1/1.2 even better) It must run on windows 2000 or earlier. It must not depend on WinHTTP. No External dependencies. You can use any open source project as long it does not depend on external dlls or files. The final executable must not be larger

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    À la une Urgent LDN
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    ...system 53 - Insurance lead generation 54 - Inbound Call Tracking 55 - Outbound Call Tracking 56 - Call Analytics integrate Asterisk call center with CRM to have some feature bellow: - Sale can use Call feature on CRM via Asterisk. - Can use feature Click to Call? Please show me the solutions (brief, draft with your paper) to continue!...

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    voip语音网关模块 S'est terminé left

    具备sip协议,支持30路以上fxs,可以实现点对点通信,需要提供相关技术支持

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    Issabel Features S'est terminé left

    We Need somene that have some skills over ISSABEL Platform VOIP , Asterisk and Call Center Platform . We need to implemented some news features into reports database .

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    asterisk ari recording S'est terminé left

    I want to create a ARI program that will play a message from URL in cloud and record the sentence spoken in channel and repeat what the person spoke after record is done Key thing here is I do not want any button pressing...Using ARI functions code should detetc when a person has spoken and when there is silence of more than 2 secs , consider the recording done Ideally i would like to implement t...

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    Rebranding of Linphone S'est terminé left

    I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source

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    ...and the VPN connects successfully, however when trying to connect through our DMZ / Virtual IP external to the internet we receieve a TLS handshake error or TLS Error: cannot locate HMAC in incoming packet Fatal TLS error (check_tls_errors_co), restarting if you think you have the expertise to help us out please let post a reply of how much you would

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    Ringless Voicemail S'est terminé left

    ...without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples [se connecter pour voir l'URL] [se connecter pour voir l'URL] [se connecter pour voir l'URL] https://www

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    Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Soft Switch development. S'est terminé left

    Hi Dear Sir How Are You. Dear Sir I want to develop a soft switch from Asterisk PBX. Please Contact me and Give me your contact number Thanks [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Soft Switch development S'est terminé left

    Asterisk based Soft Switch [Removed by Freelancer.com Admin - please see Section 13 of our Terms and Conditions]

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    Atualização do site, segue as atividades| 0.1. Conversor de tabelas para Rocket (central asterisk) | 0.2 Integração entre o Rocket (central asterisk) e PROSPECTS (area administrativa) | 0.3 Inclusao do Te Ligamos (click to call) e botao blog no menu area cliente do site 1. Blog (do 0) | 2. Integração com Rd Station | [se connecter pour voir l...

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    Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples

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