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.
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
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 pe...
I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)
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.
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
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.
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
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
...startup centred on cloud technologies. (Voip, Sip) The name is " Caraïbes VOIP " The main color must be blue like the sky. Je démarre une startup dans la téléphonie sur IP à destinations des entreprises. Le projet est axé sur les nouvelles technologies CLOUD. (Voip, Sip) Le nom du projet est "Caraib...
...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
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
We have developed a complete workflow management software the allows clients to use the CRM that is integrated with our VOIP phone system. It is built in asterix. We need developers to maintain and manage our asterix phone system within US timezone. This will be an ongoing project. Our website is [removed by Freelancer]
Hi, my names Dan. Programs like Skype and Discord are great but I'm looking for a custom-made VOIP to use with friends and family, something that is more locked off. I will need the ability to make calls, group calls and or channels. I'd like to have a portal where I can manage members and they have the ability to also create an account. I hope I
Cisco voice gateway connection to SIP server and PSTN on E1 PRI
Need the below requirement Made for business Start with unlimited calling, unlimited conferencing, toll-free numbers, and customizable caller ID. Access more functionality later, such as texting, online meetings, even faxing, without changing systems for my clients Simple to set up and use Have your whole office up and running in a day or less. Respond easily to business fluctuations or customize...
...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
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.
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
FreePBX is hosted on Vultr and voip phones are behind SonicWall. When there is only one phone is connected to FreePBX it works fine with no issues when we try to connect second phone from the same network it cannot register with PBX and can't receive any calls. Sometimes second phone gets registered and can dial out, but if we try to call some other
I’m looking for a logo for business cards and letterhead I’m looking for some flyers so that I can advertise VoIP and SIP services the Name of the company is JT&T Communications we offer best in class SIP and VoIP. And phone system analysis design and implementation
I need you configure VoIP and internet settings.
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.
...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [увійдіть, щоб побачити URL] secret GZD397XU Host [увійдіть, щоб побачити URL] port 5060 Register String 390415541985@[увійдіть, щоб побачити URL]:GZD397XU:390415541985@[увійдіть, щоб побачити URL] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [увійдіт...
...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 and is not bundled
...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
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
I need professional help with setting up duplicate Avaya voip phone system. I need worker with real experience. I will describe current system i have. I need to buy same equipment, and i need to know what licenses i really need to get or don't need. More details will be given as we talk together. Current System: • Avaya IP Office 500 V2. Build Version
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.
... The application will feature voicemail detection function where an outbound call is placed from a 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 [увійдіть, щоб побачити URL] and sip.us. The application should
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
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
i am looking for some one who can make an android mobile app that could make a mobile call trunking application for both directions GSM 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
Implement SMS verification in VoIP softphone and php backend, create IOS and Android softphone General project description: We are running an A2Billing + Asterisk server and want to have a branded Softphone for Android and IOS. Registration and provisioning of softphones must happen with SMS verification. For the softphone we want to use the source
...1main server other all asterisk server work like VM host1 IP [увійдіть, щоб побачити URL] SIP 5060 DB NAME datastore1 host2 IP [увійдіть, щоб побачити URL] SIP 5060 DB NAME datastore2 host3 IP [увійдіть, щоб побачити URL] SIP 5060 DB NAME datastore3 host4 IP [увійдіть, щоб побачити URL] SIP 5070 DB NAME datastore4 host5 IP [увійдіть, щоб побачити URL] SIP 606...
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
We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Proficient in Lantek program (lazer tube cutting) is a must. We have ongoing work for the right programmer on a long term basis.