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Добрий день, Мені потрібно виконати інтеграцію Asterisk+Zendesk - бажано, щоб при надходженні дзвінку створювався тікет, ідентифікувався дозвонювач, і т.д. Чи є в Вас такий досвід роботи? Ми знаходимося у Києві. Бюджет та подробиці оговоримо.
Hello Our current development website is https://didicar.ca. We need to add Civic plugin to the website. [log ind for at se URL] We tried to install several times, but cannot install and found some php errors. If you have experience with civic plugin, that will be plus. If this project going well, there will be more
I need to connect CRM Vtiger with Elastix or Asterisk to make calls clicking on the CRM
Design a logo "PaintnSip" We are an art studio teaching art classes for beginners how to paint. Our students are adults and we provide a..."PaintnSip" We are an art studio teaching art classes for beginners how to paint. Our students are adults and we provide a glass of wine to each student, so we call it paint and sip. We are looking for a fun logo.
I need to connect CRM Vtiger with Elastix or Asterisk to make calls clicking on the CRM
Outbound: -Realtime Monitor - here we can see the agents realtime -Scoreboard - Overview of top 10 agents -Manage Dialer - Here i can give a boost to dialer & and the...here like 77 voicemails 120 not interested -CDR -Batch Report - here i can see how good the imported data is. Ex. data A) 20 sales Data B) 10 sales -Agent Report Settings: -Sip Trunk
To duplicate an existing Asterisk Server with WebRTC ( HTTP5 - SSL - etc.) onto another machine and make some changes. Works together with an Apache Web Server and a MySQL Data Base, on separate machines. Must function as an immediate - identical backup !
zoho's phonebridge plugin supports up to asterisk 1.4 with asterisk 1.8 partially working (only incoming calls) with modern asterisk versions (11 and up) not working at all do to (apparently ) java problem zoho's support says that those values : dialed channel, linked channel, dialing channel are NULL instead of being filled with information. we
We are a Philippines Company looking for an expert in Docker, Kubernetes and Google Cloud to help us speed up migrating 3 existing VMs (Java App, Asterisk and Postgres) to Docker Containers deployed on Kubernetes on Google Cloud. We have already containerised the Apps. We now want to accelerate deployment and to then have assistance with "peeling out"
I'm looking for a tech who already has completed a Ringless Voicemail drop system. We are US based company and will target users in US only so will calling 10-digit US phone numbers. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. If you Already have completed a ringless voicemail drop system please bid
...VoIP provider. I gave the green color for your easier to understand. -- Executing [s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+918820094576") in new stack [log ind for at se URL]: Caller ID nam...
...[s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+918820094576") in new stack [log ind for at se URL]: Caller ID name is '+918820094576' number is '8902050098' This should be a...
...checkout page the fields, Pais, Endreço. Cidade,Estado, CEP, they are [log ind for at se URL] I do not know why the red asterisk does not appear. I installed the plugin WooCommerce Checkout Field Editor, although I enter the fields as mandatory, the red asterisk does not appear , and it is not possible cancel the writing (OPTIONAL). mysite: [log ind for at se URL]
Hi, We are searching for skilled Asterisk developers.
Looking for an experienced contractor to update configuration on our VOIP environment (a dozen or so phones). Currently using PiaF / FreePBX but happy to change to another distro if needed. Most phone are Mitel/Aastra 6739i and plus a couple cheaper aastra and a polycom conf phone. Key Issues / Targets: 1. Hot-desking - Configuration to allow for user speeddials to move with users as they log i...
looking to learn web scraping and accessing apis from the basics and up. MUST BE FLUENT IN ENGLISH
...retire our incoming ISDN lines and are setting up to test sip lines. We have an unusual router (peplink) and multiple redundant internet connections. We have spend many hours trying to setup our router to enable SIP connectivity however without success. We are looking for someone with 3CX, SIP and good networking /router skills. Hourly rate to be discussed
Hi Kristen H.,We would like to hire you to prepare a new catalog of @ 170 products currently on a GSA contract to mirror and add to GSA Advantage through the SIP database program. All files will be supplied to you. You will need to organize in the correct format and upload both product descriptions and photos.
Hello, We need to add a functionality to our IVR which is based on Asterisk V 13.14.0 / PhpAGI. Os is Debian 8, database is MySql 5, Php is also 5. Simple functionality: - Inbound call accepted (client who needs support) - IVR (PhpAGI) says "welcome" - Call is forwarded to 1st level agent (already done by DIAL command) - 1st level agent takes call
I have approximately 10 recordings of interviews (phone or skype) ranging from 30-45 minutes. I want these audios cleaned up to remove noise and make them sound as professional as possible. Please note there is no need to edit the actual conversation..I will do that...the only job is to clean background noise and make the sound as clear as possible
...Ukraine; - desirable - experience in setting up and administering CRM-systems for at least 1 year; - Required knowledge of php, mySQL from 1 year; - knowledge of Linux is mandatory; - knowledge level in PHP - Software Engineer, Junior level programmer will not work. Duties: - implementation of CRM projects based on VTiger; -
...years of experience preferred, US, UK, Irish, Canadian and Australian native speakers preferred, Should also have neutral accent, Long-term lessons, once a week for 60 minutes, Skype lessons preferred, My English level: intermediate to advanced, Please provide me with your resume, Should have an academic background, preferably in social sciences and economics
...(landline 1) and I have my own server too. I'd like to redirect calls made to landline 1 to another landline (landline 2) that I don't own. I've already made a vocal robot on Asterisk so that depending on the digit the caller presses, it does something different. What I need now is the last part : 1) Depending on the digit pressed, forward the call to a
We need an HTML based SIP client that can be designed to look and act like a in home intercom. For example, there should be buttons for rooms, that will let you page the rooms, and select either video or audio. This will have to be set up that each "station" can be configured which rooms it can page etc... There are a lot more features and customization
App to register with my Asterisk Server as a SIP extension. My Server will send VoIP calls to the App and the App will make a local call on the GSM network and path both calls together. In other words, the Android Phone will act as a VoIP / GSM gateway. Thamk you.
We require someone hands on experience in installing goautodial or vicidial solution on Google cloud computing. Good understanding of Linux, Asterisk and Vicidial is essential. We wish to start testing Asterisk and Vicidial in the cloud from Google. We require setup, testing and ongoing support. Phase one is setup so we can start testing. Phase
...outsource for those tasks when needed. Currently i can be specific on a project which you can tell me you can help me on this or not. We have some raspberry pi products which asterisk is installed. And there are 3G or 4G usb modems on them. Those asterisks receive calls with IAX trunking and route calls to mobile phones which are matched with raspberry
Objetivo: Provisionar teléfonos Cisco 7911G para plataforma SIP abierta (Voipswitch). Requerimiento: 1 - Selección de firmware compatible con SIP no propietario. 2 - Creación de "[log ind for at se URL]" para provisionamiento remoto.
I am having problems getting a dhadi/Asterisk/POTS configuration ( based on an old Zaptel/Asterisk configuration that does work ) to work properly. Config files and CLI output: [log ind for at se URL] Dialplan: [log ind for at se URL] The machine once finished will backup and replace my old Asterisk servers that basically operate as POTS
Hallo, I am currently learning German and I want to find a native speaker to help me to improve speaking skill. Please contact me if you have experiences in teaching German and have good knowledge about German culture and people. Many thanks and best regards!