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    81 sipura freebpx pekerjaan ditemukan, seharga USD

    Setup FreePBX on Hetzner with instructions and make two phones talk to each other.

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    I have a freepbx 15 box that WAS on a DADHI channel. I need to move it to my SIP Voip provider and have another working (freebpx 13) box that is configured for my provider as an example.

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    i have freebpx installed and i want to connect my Custom build crm with it, i need to send a parameter from freebpx to my crm and cause a popup to open so that my team do a simple form fill out.

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    I have the latest ASTERISK and FREEPBX set up on a hosted Linux machine. I am able to receive calls via the three sip trunk providers, but I am unable to make outbound calls on these sip lines. I need someone to simply diagnose the settings on the FreePBX interface, and adjust them so that they can dial out. For this, you will be provided a l...Linux machine. I am able to receive calls via the three sip trunk providers, but I am unable to make outbound calls on these sip lines. I need someone to simply diagnose the settings on the FreePBX interface, and adjust them so that they can dial out. For this, you will be provided a login to the FreePbx Admin interface, where all settings are available. You MUST have previous ASTERISK and FREEBPX experience at a system administrat...

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    i have some issues and needs to be resolved asap, if you are good with asterisk and firewall please apply, need asap

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    i have a freebox asterisk setup that i need some debugging. needs to be available right away.

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    Hello, My freebpx is not connecting to asterisk and i need a expert asap to help me with it, will be via anydesk,

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    Hello, I need a voip expert to setup asterisk with didwww need it urgently now,

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    We need help to setup freebpx with the following functions: 1. Use pin code to call to outstation & oversea. 2. There are 2 different offices. Both offices are using the same range of extension numbers (6XXX). When phone is ringing in the first office, user in second office is not allow to pick up the call. 3. When a phone is ringing, nearby user can pick up the call by dialing a code with the ringing extension. 4. Paging 5. IVR

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    I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: You will probably also need to check the G729 codec configuration

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    my provider's server crashed (raid issues) and i need a find gentleman who can install freebpx / asterisk and secure it on a cloud server having ssh access...

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    Requires an inbound truck & route to be setup for Freebpx

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    i need someone to setup freebpx + asterisk on a cloud server, need asap.

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    I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: I may probably also need help with G729 codec configuration

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    I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures:

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    i did install the freebpx but i have nothing but problems, i need someone to help me with it, secure it and make sure its working to its optimal condition. .

    $16 / hr (Avg Bid)
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    Secure my freebpx Berakhir left

    i have a freebpx 15 installed (brand new install) and i think its breached, i need someone to secure it and get it ready for production

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    I need a gentleman who will install Freebpx on Centos 5, i tried myself and got lost so i need a guy who will install it, i can set it up.

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    We need a Voip billing software and FreeBPX installed and commissioned on our server.

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    Need freebpx set up on our digital ocean server and everything linked to our sip provider. We use a free softphone client.

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    i would need some statistics from our asterisk/freepbx server we need to read from AMI directly (with node.js server) i would need a statistics cronjob which run every minute and enters data gathered into a mysql table data that shall be fetched: calls in the queues (customers waiting) calls in progress (customers calling with agents) waiting time per caller in the queue Talking time per caller in the queue we have more queues so queue info must also be stored We have already installed 1)Node.js 2)pm2 already there runs a process which processes other information so the scheme is already setup- we just need an experts who extends the functionalty

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    I have a freebpx installed in a cloud server and i always have problems with a error, i need a guy who can help me with my setup...

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    Integrating plivo with freebpx / asterisk to build switchboard support simbox and hotline

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    EducaCallcenter Berakhir left

    Tích hợp plivo vá»›i freebpx/asterisk để xây dá»±ng tổng Ä‘ài support simbox và hotline

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    Hi Viktor S., I noticed your profile i install freebpx on raspberry this my first time its connected with my phone by bluetooth i want forward income calls to go by my mobile

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    Hi Pavel E., I noticed your profile and would like to offer you my project. i install freebpx on raspberry this my first time its connected with my phone by bluetoth i want forword income calls to go by my mobile

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    Hi Anton F., Ii install freebpx on raspberry this my first time its connected with my phone by bluetoth i want forword income calls to go by my mobile can you work on that

    $25 - $25 / hr
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    0 penawaran

    installing asterisk and configuring needs

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    Hello, i am looking for a individual who can help me with a freebpx setup, i have one cloud server that i need ocasionally help with, setting up trunks, extensions and troubleshoot, Let me know.

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    Hi, I have a need for an app which can connect to my Freebpx/Asterisk Server I will give further information to any replies.

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    I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.

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    Lokal
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    I am looking for an expert programmer in asterisk that can set all call rejection codes to 503 and Instantly reject all busy circuit calls without any delay by removing the played busy circuit massage, these mods needs to be done on 2 Freepbx Server running Asterisk 1.8.17.0

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    I need my Freebpx restore on my computer asap

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    Hello, i need a event based asterisk, freebpx and A2billing expert, i have a single box that sometimes i get stuck and need a guy that can help me with it time to time, Thanks

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    ...puerto Syslog (UPD 514) y recibir todos los mensajes que envíen los distintos equipos pasarelas de Voz IP (VOIP Gateway) y procesarlos, que muestre en pantalla cuando cuelgan/descuelgan la bocina, marcan, timbra, tarifa, y las demás acciones que se presenten. El programa deberá interpretar los mensajes Syslog de los siguientes equipos como primera fase del desarrollo: - Linksys (ej. PAP2) - Sipura - Grandstream (ej. GXW 4XXX) Ademas deberá contar con base propia local (archivo) para manejar las tarifas establecidas y un pequeño reporte de llamadas (3 días). Como segunda fase de desarrollo: - Soporte con otros equipos: - Cisco - HTTEL - Integración con A2Billing - Desarrollo de un pequeño web API pa...

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    ...puerto Syslog (UPD 514) y recibir todos los mensajes que envíen los distintos equipos pasarelas de Voz IP (VOIP Gateway) y procesarlos, que muestre en pantalla cuando cuelgan/descuelgan la bocina, marcan, timbra, tarifa, y las demás acciones que se presenten. El programa deberá interpretar los mensajes Syslog de los siguientes equipos como primera fase del desarrollo: - Linksys (ej. PAP2) - Sipura - Grandstream (ej. GXW 4XXX) Ademas deberá contar con base propia local (archivo) para manejar las tarifas establecidas y un pequeño reporte de llamadas (3 días). Como segunda fase de desarrollo: - Soporte con otros equipos: - Cisco - HTTEL - Integración con A2Billing - Desarrollo de un pequeño web API pa...

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    Hello guys, This project is belong to who make are expert. i need to install FreeSwitch on my server and install ASTTP on that. then doing this configs: 1- Activate G729 Codec on that 2- Change transfer protocol to UDP 3- activate just SIP protocol on server. (disable another protocols) 4- Enable Video Call feature on freeswich if you not robot please add "Swh" in your bid. Thanks for your attention

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    HOJE O SISTEMA ESTA CONFIGURADO COM AS INFORMAÇÕES DESSE SITE: COLOCAMOS AS CONFIGURAÇÕES LOCAIS DE CADENCIA E DISCONECT DESSE SITE: PROBLEMA: AO RECEBER UMA CHAMADA, APOS DESLIGALA O SP3000 NÃO DESLIGA A LINHA, SE ALGUEM TENTAR LIGAR PARA LINHA DA ERRO E SE ALGUEM TENTA LIGAR DA LINHA DA OCUPADO ATÉ QUE REINICIE O APARELHO, NOTAS ISSO SO OCORRE NA RECEPÇÃO DA CHAMADA, AO EFETUAR A CHAMADA E DESLIGA A LINHA É DESLIGADA E PODE SER OPERADA NOVAMENTE, OBS. NO MOMENTO QUE TRAVA O DISPOSITIVO FICA COM TODOS OS LEDS ACESSOS ATE REINICIAR. PRECISO DE UMA SOLUÇÃO PARA ISSO ALEM DE ENCONTRAR UM SISTEMA DE RESTAURAÇÃO DAS CONFIGURAÇÕES

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    Freebpx Issue Berakhir left

    issue in Freebpx

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    Configure Freebpx Berakhir left

    We have UP & running Freebpx(VOIP solution),i need to merge two SIP in a single Freebpx.

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    ...be presented with the option of using current number or enter in a new number (10 numbers only: 0NXXXXXXXX) giving estimate time of callback. Once the telephone number is entered it will remain in queue until it's turn in Queue. (queue show XX) Once the customer is next in the queue will transfer it to the next available Agent and play an announcement to Agent, then call customer. (Use FreeBPX CIDLookup on outgoing number and append to CID if no lookup returned show calling number in CID) More than one Queue can have this option. Freepbx 13 Module Two “Queue Callback Report” Reporting Portal All Callbacks Lodged (Dates/Times, Queue, Number, Current Position(Real Time)) marked as Pending or Successful-(Exten(Agent)) ...

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    Bersegel
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    Freebpx expert Berakhir left

    Hello, I am constantly setting up freebpx and maintain them, i need a expert in this feel to help me troubleshoot and help me manage the current bpx as needed.

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    I have a Juniper Networks SSG5 Router. I also have VOIP service. I need my router programed to allow the Sipura 2100 box to communicate from behind the router. It is highly likely that the programming can be done remotely or in conjunction with my assistance.

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    I have a SPA-3000 Linksys device which I want to use to: 1. Already have a voip account number 2. Need to route all incoming pstn calls to another number via VOIP

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    i am dealing with didx and there's problem connecting them with my did so i need asterisk freepbx expert help me to setup the trunk and show me how to do next time thanks

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    I need to perform a complete check on my asterisk pbx-free 1) I have a problem with the trunk that I can make calls but not receive 2) I need to configure a particular number in receipt to respond. 3) the exact configuration of a Portek 2 sim 4) Configuration of a sipura 2002 output

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    Freepbx Install Berakhir left

    Remote configuration of freebpx. -Configure FXO ports for inbound and outbound calling. -Configure one phone for remote use (sip proxy phone). -Configure 10 extensions -Configure voicemail to email -Configure simultaneous ring (SNR) -Configure Auto Attendant per pre-determined flow chart.

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    I have setup the FreePBX Box on OpenVZ VPS at A2hosting.com. The box can receive incoming calls intermittently. It is erroring when trying to dial outbound to CallCentric SIP Service. Asterisk does not start automatically with the service, I have to start it manually. The error logs show FreePBX cannot connect to the database. I need the PBX configured and setup properly. I will need limited support moving forward.

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    We have a server online working, now need to install freepbx (others can be considerated) and configure our VOIP trunk on there.

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    1. Customize any open source Android and IPhone client. 2. Add messaging functionality and online presence status. 3. Connect both to our SIP server. Do necessary configuration on SIP server. If developers already have created their own dialler, they can use that. SIP server will be running Asterix/freeswitch/freebpx or similar. Developers must provide suggestion of which diallers they can customize. We have an application with a user database, from there users will also be able to make calls and message users with the softphone applications installed. This integration needs also to be done.

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