Asterisk pbx sbcprojekty
Configure and maintain a Sangoma SS7 SBC. It has SS7 connections with operators and a Sangoma Freeswitch setup.
Looking to set up a custom skinned rebranded VICIDIAL ISO Image for call centre application . Experience is Asterisk , Vicidial , Free PBX , Linux etc preferred.
...following guidelines: -clever/funny -photo real, not cartoony -homey/cozy feel -shows product (projects) The card “opening” should be achieved either by having an animation or utilizing scrolling in a way that delivers the punchline appropriately. The digital assets for the projects will be delivered to you and we need 2 versions of this “card”, the only difference is that one will have an asterisk in the text with a footnote that reads “gift amount in accordance with EVP reductions” or some such joke they will understand. I have attached an example of the "cozy" element we are looking for that was on a card in years past. This is time sensitive now that we are in Dec., our previous artist was a flake that put us behind schedule....
We are a telecommunication company in Turkey which mainly operates in cloud pbx area. We need a softphone that will work with our pbx systems. It needs to work on 3 platforms; IOS, Android, Windows. You can see the detailed information below; • “Sip Server – Username / Password” based login screen - (We require this for the current project. After this project we are going to need default registration method for international use) • QR Code – We need to implement QR code for easy registration and configuration steps • Config file or URL – Same logic as QR. We could use this on computers. • Video Call Support – (H264) • Admin Panel – We require this for all platforms. It will show usage statistics and we must a...
Need to upgrade existing Free PBX platform and insure redirection of its press options to connect to a sip server via webphone/softphone. Experience with call routing on selections from PBX options to sip server and interconnectivity thru voip(softphone) and pstn are required. Knowledge of Javascript and API implementations(Documentation is available) are required to hand-shake Mizutech webphone software with the FreePBX. Migration of files from external hosted server to the PBX server may be required.
Hi Mohammad, I noticed your profile and would like to talk with you about one possible Project related to one PBX Elastix. The question is: Is it possible to add to Elastix PBX extensions working over wattsApp calls? If yest, watt is your price and requiremants to do it? Best Regards
i need someone who can design a web-based sip phone. which can communicate with any SIP server like an asterisk, freePBX, VOIP, SIP Server, and the browser should support firefox, explorer, and chrome. Also, am expecting a demo which is already done or ready one, so that will agree for implementation
Configure server kamailio with modules : NAT TLS Integration MS Teams Register SIP User Local. Options Forwarding SIP Register to Asterisk Call Routing, OUTGOING. Softphone -> Kamailio( Integration Asterisk) -> MS Tems 365 INBOUND MS Teams -> Kamailio( Integration Asterisk) -> Softphone Softphone -> Kamailio( Integration Asterisk) -> MS Teams 365 Skills: VoIP, Asterisk PBX, Linux
We have Yeastar IP PBX S20 installed and now we want to connect GSM gateway Yeastar TG-200 (2 sim) with this PBX to manage mobile calls. We need you to do configuration so that we can manage / divert mobile calls to different extensions. The quality of the voice should be as clear as if it is dialed directly on the mobile.
To invoice clients and credit agents with none standard Invoices and Credit-Notes. See the 3 attachments for more information. --- Please note that You must be on this site to answer, because of bugs in the system.
Preciso integrar 1 tronco de meu pabx atual com o CRM da AMO CRM Segue documentação necessária, necessito de uma certa urgência obrigado
I need some help decoding/encoding ASN.1 in Python, Specifically using pycrate to encode/decode LTE SBc messages, Is this something you can assist with? Thanks, Nick
Hi Zhang X, I need some help decoding/encoding ASN.1 in Python, Specifically using pycrate to encode/decode LTE SBc messages, Is this something you can assist with? Thanks, Nick
Efetuar a configuração de um servidor asterisk 16 que tenho na nuvem (CentOS 7) com ramais SIP Polycom, integrar Asterisk com SQL local, dashboard do Asterisk mostrando usuários logados usando API.
...Dec 04, 2020 We have a project which is a Help Call system for the elderly. It's basically a fancy Voip phone running Openwrt and Asterisk on the MediaTek MT7688 CPU. We are using a WM8960 codec connected via i2S to the CPU and we are performing some basic audio functions to check the audio (record & play) using arecord and aplay. We are getting some odd results such as noise being introduced and fast garbled recording. We suspect issues with the audio driver. It is possible some of the registers on the codec are being messed up. We are looking for some direction as we need to solve the audio driver issues before we get the Voip calls moving through Asterisk. Our first task is simply to play and record audio cleanly using arecord & aplay. Any assistance/d...
We have a requirement to fine tune a VICIAL configuration consisting in the following setup: * HP Proliant DL380 server with 64GB of RAM with a planned upgrade to 128 GB HDD 1.2TB SSD 800GB This hardware has installed VMware 10.7 on it and Asterisk + VICIDIAL instances are being created on it This setup should allow more than 250 simultaneous calls but currently can't handle more than 200 so we need to know where is the problem. In addition we would like someone who can do maintenance remotely via a Windows Desktop in Amazon EC2
Hello, in the past we had an Elastix central, with a callback function, which we used as follows: <extension>&number= <number_to_dial> When you clicked on the link, the extension started calling that phone number. The issue is that we migrated from Elastix to a Grandstream UCM6200 PBX and we lost this feature. Is it possible to have it again with this hardware? Experts only please. Thank you.
I am looking for a Delphi developer with the following Skill sets. Fusion PBX Voip client applications on Iphone and Android. I have a Fusion PBX server set up that will need modifications & maintenance. I have an IOS & Android app that I am having issues with the app in the background. ( Android )
I am looking for a Delphi developer with the following Skill sets. Fusion PBX Voip client applications on Iphone and Android. I have a Fusion PBX server set up that will need modifications & maintenance. I have an IOS & Android app that I am having issues with the app in the background. ( Android )
je veut mettre 2 serveur pbx derriere 2 routeur mikrotik ccr1009 avec vrrp. je veut configurer les pbx avec des ip public avec le mikrotik comme gateway pour permettre de filtrer et bloquer avec le firewall. je veut mettre les mikrotik en mode transparent, comme on peut faire avec les firewall sonicwall. voici le plan. I want to put 2 pbx server behind 2 mikrotik ccr1009 router with vrrp. I want to configure the pbx with public ip with the mikrotik as a gateway to allow filtering and blocking with the firewall. I want to put the mikrotiks in transparent mode, as we can do with sonicwall firewalls. here is the plan.
Small office with 4 analog incoming lines and 10 extensions. Currently using UCM 6204 on which extensions are setup. Need a solution for accessing logs of: 1- Missed calls 2- Incoming calls 3- Outgoing calls 4- SMS to be sent to missed calls only Be able to access the system from local lan
DetailsProposals Project Details Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File: Range;country;Number;Carrier Payout;Carrier Pay Term;Client Payout;Notes (remarks) -Numbers should be all routed to a local IVR (or a group of IVRs, which ...for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" including Asterisk/FS Skills Required PHP JavaScript Software Architecture ...
i need to integrate issabel pbx with grandstream ucm
I have a freepbx system running at the office. I would like to have off network phones connect. I have one off network phone that works but when I put that same phone behind a cradlepoint (verizon) router I cannot connect to the freepbx any longer. So I need help configuring the cradle point as the phone is correct setup and the free pbx along with the router at the office is good it comes down to the cradlepoint.
I have a freepbx system running at the office. I would like to have off network phones connect. I have one off network phone that works but when I put that same phone behind a cradlepoint (verizon) router I cannot connect to the freepbx any longer. So I need help configuring the cradle point as the phone is correct setup and the free pbx along with the router at the office is good it comes down to the cradlepoint.
We are looking to use the ASTPP software for our small wholesale voip business and possibly the IP PBX as well. Many thanks Doug budget varies depending on time and ongoing support considerations.
To invoice clients for incoming calls and credit agents for work. Please see attached file for more info.
I'm looking for someone who can develop an integration between Asterisk (v.18) and AWS Amazon Polly. Call flow: 1) Asterisk will answer the call 2) Amazon Polly will greet the caller - part of the text is fixed, a small portion is dynamic 3) That's it. There will be no interaction (voicebot). Asterisk will answer, Polly will greet and byebye. This machine is running under Debian.
Preciso de uma api que quando a ligação entrar no queue e ramal que vai atender tirar do gancho ele disparar uma agi.
I have asterisk pbx system and I am looking to update it by adding a phone application. for example, if the phone is not answered in 3 rings it starts ringing in the phone app.
I need some help with selling something.
I need a experienced one to develop an Asterisk Open CTI connector with Salesforce CRM. SalesForce Lightning Platform Starter requirements - Enterprise Edition.
...ability to run asterisk 1.4 on the Community Edition of Clearos. the OS can be found (with the MARIADB installed) Asterisk files can be found: https://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.11.1+ Asterisk should be compiled with the following options files ( and ) NOTE: res_crypto MUST be included as a requirement Deliverables must include a step by step reproduce-able solution that compiles and allows the Asterisk to be run from the asterisk -cvvvvg command
create custom module in prefex crm integration with calls report and recording popup up screen when incoming calls and appear customer details or add new . click to call from crm . Initiate calls right from CRM. Instant pop-up alerts for incoming calls Single click dialing for outgoing calls. Integration logs both incoming and outgoing call. Complete conversation history in one unified window. Thanks
Deliverable - successful integration, code, documentation and training Integration functions: Develop API for integration between Cloud PBX and CRM Make calls from CRM interface in one click (Desktop App, Browser, App) call statistics integration with call recordings on CRM Callback features Please provide following details when you are quoting price 1. How many years of Experience you have in Third party API Development 2. Please provide description of APIs developed by you. Please specify telephony integration APIs along with others. 3. How many year support you will provide for API developed. Any bugs that arises need to be fixed
Hi I have a TB 3 (waffle) and I am setting it up with a raspberry pi 3 SBC . The laptop is ubuntu 16.04 Most of the system is up and running. I need some help on an hourly basis to get this working by using Skype and team viewer. I only want people that are experienced in ROS to apply. I think it will take 1 max 2 hours to fix and then I may want some ongoing ROS help If you can not fix this please do not waste my and your time. I prefer people on a USA time zone You will need Linux, ROS OpenCR Raspberry skills you get paid only if the problem is solved $20 per hour is fine for this work.
Configure server kamailio with modules : NAT TLS Integration MS Teams Register SIP User Local. Options Forwarding SIP Register to Asterisk Call Routing, OUTGOING. Softphone -> Kamailio( Integration Asterisk) -> MS Tems 365 INBOUND MS Teams -> Kamailio( Integration Asterisk) -> Softphone Softphone -> Kamailio( Integration Asterisk) -> MS Tems 365
We need to take the well-supported BeagleBone Blue design and modify the layout to a "pressure tolerant electronics" architecture. This will entail a number of mechanical considerations (e.g., SMD connectors become through holes), and changes in packaging, but the board size can be expanded as necessary. We use Eagle and SolidWorks, but as long as we can get compatible files in the end...architecture. This will entail a number of mechanical considerations (e.g., SMD connectors become through holes), and changes in packaging, but the board size can be expanded as necessary. We use Eagle and SolidWorks, but as long as we can get compatible files in the end, any platform can be used for the work. We'd like a ballpark estimate for a job of this magnitude, i.e., the re-layout ...
...years with full computers or tablets and they have easily found ways to power them/turn on/off. Being on the hardware side of technology I'm sure you know all about single-board computers (SBC). We know they are the future of computers and they are ALWAYS getting better. Trying to add an SBC to a car has problems when it comes to power tho. They all have different connectors (USB-C, micro USB, barrel jack, etc) different voltages (3,5,12, even 24), and different OS's (Android, Windows, Linux, etc.) With so many different things to worry about it's hard to find a power supply for a car that will turn off/on the SBC when the car turns off/on. I believe I have found a way to power all of them and have them turn off/on with the car. What I'm looking ...
I have a CRM created with php codeigniter framwork, I need to start calling via this crm (making outboud calls), so by connecting it to an asterisk system and a sip account.
Hi Viktor S., are you expert in troubleshooting a "toll fraud" attack on our asterisk/freepbx server?
Hi Anton F., are you expert in troubleshooting a "toll fraud" attack on our asterisk/freepbx server?
we need to set multiple sub-domains ie multiple companies/customer under single installation of FusionPBX. Need to access admin/users/ext of each of this sub-domain/tenant. Need help to set this up.
Need to Install Fusion PBX for press 1 survey campaign and IVR broadcasting
ATI is looking for development, implementation(s) and support of Asterisk, Flexisip, Turn/Stun, Linphone and Library repository (IOS only). Phase I will be slight modification and branding of the GUI and App. Setting up the core on our test servers, placing the App in the App store and testing the total solution. Phase II will involve deploying the solution at multiple customer sites and programming specific to each client. Phase III will include further modification of the App and building a management console (node.js) that will be deployed at each location. Phase IV is continued support, modification’s and future deployments at new customer sites. Key components: • All services and applications running on our client’s servers and network. • Setup of cli...
...release the call and continue the transfer - x202 still sees "201" as Caller ID, until he picks up the call. Once he picks up the call, 222-222-3333 appears on the CID. I want to change the way the last step in the above scenario works. Once x201 presses Transfer again to release the call and continue the transfer, x202 should see 222-222-3333 on his CID display. This worked for us when we used asterisk. In , setting connected_line_method=update made this work correctly. If anyone has the right experience, I will be happy to hire you for this project. I do expect that you already have the experience and not wait to do the research after you are given the job. If you already know how to get this done, I don't expect to spend more than an hour with you working on ...
Currently we have several PBXs in Asterisk and there is a problem of Flooding of port 8089 Webrtc, it is not an attack since the traffic is valid and comes from IPs of our clients, this problem happens when the person who is using the Webphone has intermittence of internet, generating multiple connections in closing state that are observed in the Kernel log. We think that to solve this is to use an Opensips such as Webrtc Gateway and that the flooding is controlled from this point. We need a professional to help us install and configure an Opensips as a mid-registar WSS, which allows us to log the extensions found in Asterisk of the SIP type. Observation: The asterisk actualy is not Real Time, and for local devolpment with cannot change that.
Need to Confgure Goautodial V4 for a press 1 survey campaign and a IVR Blaster Campaign. need to configure dynamic login to self authenticate and stop sip attacks on the server
I m working in voip in Morocco from 2014 I have a system of asterisk Isabel the problem is when my client send me trafic give him an erreur 486 must be 503 In this case my client lost his traffic https://www.freelancer.com/users/l.php?url=https:%2F%%2Fusers%%3Furl%3Dhttp:%252F%%252Fpt%252FA42hBUju%26sig%3D50c67e1b3a7f27f20771fd2ee320f07b9eb302904f9e19edea3a28d328aafbdf&sig=dbffe148af4b5367f232a97a6acda81aea4364dfabcb09ccf07b413f3f1c3be9
