Per la nostra azienda disponiamo di un centralino Asterisk installato su una vps. Sto cercando una persona che possa intervenire ogni tanto per eventuali modifiche alle funzioni del centralino. Vorrei stabilire una tariffa oraria per gli interventi che potranno essere fatti da remoto.
Buonasera, dobbiamo risolvere il problema di interfacciare una nostra applicazione gestionale Web con Asterisk, in modo che, in primo luogo, si possa chiamare un numero di telefono dall'applicazione (in secondo luogo innescare un popup all'arrivo di una telefonata). E' disponibile per una consulenza? In caso, il mio indirizzo di posta è [login to view URL] at
Good morning, my name is Fabiano and I need your help. I need to install Asterisk + FreePBX + FO2 panel on a vps. distribution choice centos 5.9 32 or 64bit centos 6.6 32 or 64bit centos 7 64bit centos 7.1 64bit Fabiano
I have posted this project before and many did not understand what they were bidding for. If you are interested, Please ...issue a hangup on sip side, i believe sip 487 Sip client on phone 1 will only register to sip server when it detects that headset jack is being used. If you can also make asterisk work as phone 1 (using chan_alsa) Please mention.
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 22.214.171.124
i got this error in asterisk and freepbx under centos "The number you dialed is not in service, please check your number and try again" im using twilio sip trunk, please i need soeone who fixed, everything i twilio is already done
I need a quick https server in delphi 2006 using clever internet components. they have an httpserver. component and support for tls 1.2 Would like it enhanced to load a self signed .cer an d key and accept a request. ASAP
Zoho's phonebridge plugin supports up to Asterisk 1.4 [login to view URL] With modern asterisk versions (11 and up) it's not working at all do to (apparently) java problem. And the property in the phonebridge adapter seems like it is not being set properly. With Astersik 1.8 it was working partially
...on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud
...page is to be seen or the program is to terminate. In addition, parts that have quantity on hand values that are equal to or below the reorder point should be flagged with an asterisk. Write this program as a C++ program using structures that have bound methods, functions. Write a structure, **struct card**, that will represent a card in a standard deck
...Main goals to be meet : - ESP8266 need to connect to local wifi - able to establish the connection as a mqtt client to a cloud based mqtt broker (a [login to view URL] file provided for TLS encryption, n broker is additionally password protected ) - continuously send a sensor data to mqtt broker on a fixed interval of 5 sec - subscribe to a topic and on receiving
A company is running Asterisk with an onboarding system. We are looking for an engineer who can support Asterisk and Kamalieo system in the cloud. Engineer is required to set up an onboarding system for clients and support them on daily basis. Here is the following job scope. 1) To set up and maintain an onboarding system that allows a user to sign
hello, we recently had some work done by a freelancer on our bespoke asterisk voicemail application. however, customers are reporting various problems: connectivity, hang-ups, can't make changes, etc, etc.. : something is definitely wrong with what was done. we need an experienced troubleshooter to take a look and make changes to a live, working system
hello i want to make a dynamic filter with asterisk follow this features - the calls come from customer pass through asterisk filter to check if it's an spam call before to send to gateway gsm if the call is a spam, then the call no go to the gateway but if the call is a real, then the call pass normaly
Hi I need someone who is diligent with Voip Asterisk. We have a Voip freepbx setup on 1.8 For some reason Iptables is blocking our Phone lines. I disable iptables service. All is working 30 min later it automaticaly reenabled iptables no more Phone. So as the freepbx is old. I want to reinstall a new server. We have 2 tenants 5 trunks and 11 phones
...programmer that has years of experience with the language and also has allot of experience with SMTP/Bulk mailing applications and all the SMTP email security protocols such as SSL/TLS. Programmer must know how to do multi threading and also know how direct mailing works (sending emails through DNS without any SMTP). Must also be able to implement everything
...person to do some kamailio development for us. We would like to have kamailio look up the registrar domain and forward all registrations and invites to and from multiple asterisk servers. I think this can be done with Domain and Dispatcher module. I have written some of the config to handle registration, but when a call(INVITE) comes in it is not
I offer VOIP telephony services and development of telecommunication solutions, I need a logo and a brochure for my site currently Im using logo which is n...logo and a brochure for my site currently Im using logo which is not mine. Send sample on your propoersal based on the description of my needs This is my site [login to view URL]
We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.
Need to configure 10 internal accounts on IP phones, connect 3 external telephone lines. Set the voice greeting and the voice menu according to the technical task. Implement a CDR report and record conversations.
We would like to install a fresh FusionPBX on VPS server what we need it secure: a. Firewall b. SSL / TLS (Let’s Encrypt) c. XML RPC d. Fail2ban Do test make sure everything works for multitenent clients etc. . Please message me for more inquiry that i need. Thank you.
I have 2 asterisk (FreePbx) servers up and running in 2 different locations connected with an iax2 trunk. I need help with the following: 1. Install and configure chan dongle for use with Huawei K3520 in one of the locations. 2. Dial plan between the 2 asterisk servers, depending on dialed number the call should be sent to correct dongle. I have a
I need an application that runs on Android phones (4.0 or above) and can send/receive calls 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. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please let me ...
Fix an existing Asterisk PBX system with WebRTC. Right now calls don't come in because FreePBX/Asterisk doesn't register with DID supplier. To make it register, some changes should be made. The system consists of 3 servers, Apache, Asterisk and MySQL. You must be able to work with Teamviewer !
...wondering if you could make a modification for me. It seems that Etsy now require TLS V1.2 for REST requests and the program does not work with this. If you could fix it for me, let me know how much? I have tried, but I can't seem to get Delphi's REST Client / REST Request to use TLS V1.2. I have modified the original program slightly, so I will need to send
In Need for SSTP/TLS VPN Client for andoid 1. create Profile 2. Add certificate server certificate must be signed by CA or Server Client must match CA imported 3. Add default route or add speficic IPS/DNS to be routed through this tunnel 4. Start in startup option 5. Icon /run in background 6. Auto reconnect /save password option 7. enable mschap
We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : [login to view URL] regards
Create and train italian acoustic model based on CMU sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should...on CMU sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should be configured for telephone (8 khz), in order to integrate it in Asterisk with mrcp server.
...a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows. The server side it's done and use Asterisk server with PJSIP. I would like that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features in the 1st step of the project:
...someone to handle the very few tech support calls I may get while out of town. Might go the whole week with none sometimes. You need to be very knowledgeable of Freepbx and asterisk in order to support my clients. I will on occasion need projects handled such as repartitioning, adding a logical volume etc... Trust is a key because I will be giving you
...streaming HLS and HTTP streaming support Security Features: Hotlink protection (restriction of HTTP referrer) Secure Token (secured URL) Shared SSL (TLS) Custom SSL (TLS) Let's Encrypt SSL (TLS) OCSP stapling Block bad bots DDoS protection Pull Features Push Features: Instantly purge zone caches or single URLs - Upload content to your FTP