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    Softphone android Berakhir left

    Un softphone que maneje cuentas SIP para ANDROID

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    Saya Pernah Direkrut Ngentry Dikantor Badan Stastistik Itu Terjadi Tahun 2015 . Jam Ngentrynya Dimulai Jam 12 Siang Sampai Jam 4 Pagi , Kebetulan Saya Sip Jam 9 Malam Sampai Jam 4 Pagi .

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    setting IP PBX Berakhir left

    jasa setting untuk keperluan kami yaitu...yaitu: Billing System / Budgeting per ID / User Recording All Call IVR / Auto Attendance Call Fax to Mail & Mail to Fax Bisa Cabang IP PBX Yealink BLF [Busy Lamp Field] Sip Messenger connect Android APPS VOIP saat ini kami memakai program elastix dengan hardware openvox a814hj0582xk dengan chip 8fxo

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    ...Care untuk bekerjasama dengan PT Kronus Indonesia - Perawat, Fisioterapi, Okupasi Terapi, Terapi Wicara & Bidan - Laki-laki dan Perempuan - Usia 21th – 45th - Memiliki STR/SIP yang masih berlaku - Mampu bekerjasama dengan Team / Independent - Penempatan wilayah Jakarta dan Luar Jakarta (Bandung, Semarang, Yogyakarta, Surabaya, Medan, Makassar atau

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    this program for information tcooperation in Goverment "Pemerentah kabupaten WONOGIRI"(Dinas Koperasi dan UMKM) INDONESIA, Can use for ALL Dinas koperasi and UMKM ind Indonesia

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    We are developing a telephony to chatbot tool that uses the open source Voximal Suite [login untuk melihat URL] with our Pandorabots chatbot https://home.pandorabots....take over the phone calls with the customers. In that regard we are seeking assistance in integrating a capability where the customer service agent can take over the call via SIP (VOIP).

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    Hi Sunil I saw that you have experience with SIP and VoIP, on android and iOS I need someone to build iOS and android App. If you are interested we can continue talking on chat See you

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    Hello There i need some help to make some config on my FreeBPX server like: 1-Remove and add new SIP line 2- there is problem to login to the control panel after i tried to make restore backup . so can you do it for me and home many time you need to finish it note: there is teamviwer access and then you can access the Unix server by web

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    Hello There i need some help to make some config on my FreeBPX server like: 1-Remove and add new SIP line 2-make my system cluod on internet so the agent can use there extension in anywhere 3- there is problem to login to the control panel after i tried to make restore backup . so can you do it for me and home many time you need to finish it

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    Skype SIP client in PHP 4 Hari left
    TERVERIFIKASI

    We have a few Skype Connect profiles with inbound numbers at...script should (persistent) connect to Skype servers, login and answer the calls by returning a recording (depending on the inbound number). That's it folks:-) It's all standard SIP, pretty straight forward. --> $100 FIXED (it's max 3 hours work, MAX, 30 minutes if you've done it before)

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    Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

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    SIP register with Custom software 1 hari left
    TERVERIFIKASI

    hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [login untuk melihat URL] i want to run this program on a router Router Link [login untuk melihat URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580

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    Programming Bandwith.com 1 hari left
    TERVERIFIKASI

    Help programming sip endpoints, applications voice & sms in [login untuk melihat URL] I need to connect my [login untuk melihat URL] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [login untuk melihat URL] Prior experience

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    Multichannel Phone Dialer 1 hari left
    TERVERIFIKASI

    In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number

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    I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

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    sip to Native dialer APP 16 jam left
    TERVERIFIKASI

    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this

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    i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

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    VOIP, SIP Expert 6 jam left
    TERVERIFIKASI

    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

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    I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

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    i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

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    you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

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    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

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    ...com/click-to-call/ etc You don’t even need a fax machine…,Receive Fax from customer as PDF in your Email. The system will support setting up/administrate the call center SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability to activate/deactivate service

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    Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable

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    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

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    I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

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    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

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    I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.

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    ...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me

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    I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.

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    WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network

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    VOIP ASSISTANCE Berakhir left

    VOIP ASSISTANCE SIP DEVELOPPEMENT

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    1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.

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    ...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [login untuk melihat URL] secret GZD397XU Host [login untuk melihat URL] port 5060 Register String 390415541985@[login untuk melihat URL]:GZD397XU:390415541985@[login untuk melihat URL] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [login untuk melihat...

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    ...to amend for us. Do you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications. Zoiper softphone is a standalone client-side software VOIP phone application

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    ...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new

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    Lightweight SIP Client with Dynamic CRM which would display the customer information. CRM Field and information would display dynamically and action should collect and send to server Database. Required Features: SIP Call with transfer/conference/hold/Mute facility SIP messenger. Dynamic CRM-Dynamic field header and value display for each call according

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    I need some one who is expert in cyber security & hacking and specially VOIP, Sip, you should be expert in this field.

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    I need professional help with setting up duplicate ...• 18 Avaya Phones – Model 1416 • 23-24 extensions Needed • 4 Trunk Cards That I see • 5 Analog Phone Numbers Following Licenses currently on system: • SIP Trunk Channels • SIP Trunk Channels • SIP Trunk Channels • Essential Edition • Essential Edition Additional Voicemail ...

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    as the title indicates, need sip installed on cisco 7960 to work with astersik. total 10 phones

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    Hi. We need to configure asterisk on our ubuntu machine. We need someone high skilled to do the job. We will also need to make it visible over internet with public IP. Also we need to configure some IP phones on it. After all, we need a manual with step by step about how to configure it.

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    I have lan calls working fine, but require someone with Cisco/Asterisk SIP knowledge to help resolve this issue with NAT, no audio not working.

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    I require a voicemail d...csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [login untuk melihat URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

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    Predictive Dialer Berakhir left

    we need a Predictive Dialler with freepbx to link to Qs and have a wall-board with reports. reports need to link to recordings. this all has to work on freepbx and sip and softphones. The PBX will make the call and the agent when signed into the Q will receive a call. the agent phone rings first then the PBX makes a call according to a predefined list

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    Hi I need an application that can receive a call through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset and set it as default

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    ...to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work for both and each by separate registration. 4- send calls from sip/iax2 to gsm and receive calls from gsm and send it to sip/iax2 5-

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    ...provisioning by email address login. Users should instead enter their mobile phone number to register. After receiving an SMS with a verification code the softphone should download the SIP provisioning data from the server. - Create IOS version and deploy to appstore - Create Android version and deploy to playstore Server side works - Install vialer-middleware

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    ...1main server other all asterisk server work like VM host1 IP [login untuk melihat URL] SIP 5060 DB NAME datastore1 host2 IP [login untuk melihat URL] SIP 5060 DB NAME datastore2 host3 IP [login untuk melihat URL] SIP 5060 DB NAME datastore3 host4 IP [login untuk melihat URL] SIP 5070 DB NAME datastore4 host5 IP [login untuk melihat URL] SIP 6060 DB NAME datas...

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    Hello All, We need to integrate Vonage and Callcentric SIP with our Bitrix24 installation for calling to USA. This will be used with a 10 PC user system for a call center. Other requirements with same: - SIP set up in Bitrix24 - Setting user right for 10 users with extendable option to more users. - Setting up rules for call forwarding, call picking

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