Need someone with openSIPS/freeswitch experience the initial task would be to help us setup and configure a high availability deploy of OpenSIPS ->[ Freeswitch1, Freeswitch2, Freeswitch3, … ] the other aspect is I want to setup OpenSIPS to handle … - [url removed, login to view] - [url removed, login to view] Where each subdomain has different ip whitelists and also help us setup graceful deploys where we can take Freeswitches out of circulation to push changes without dropping calls by draining live connections and swapping in standbys its a pretty big project
I need help to create a trunk between Freepbx and my other SIP server. There are E1 cards connected to FreePBX and we assign DIDs from Freepbx to end users. They register by ID and Password to receive incoming calls. We don't have any issue there. But now we have another request to forward bulk DIDs to another SIP server and the authentication will be only IP without any user or password. It is simple project which should not take more than 1 hour to complete for anyone who knows how to work in FreePBX.
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Hi, Looking for some one who can teach installation and customization of Asterisk freeswitch a2billing kamailio all in standalone server and connecting of writing application
I have a fresh installation of ASTPP , i need the following configured: 1. create origination carrier ( customer) 2. create termination provider [url removed, login to view] origination rate table and termination rate table 4. create trunk and set up routing so that origination carrier calls cam be terminated to termination end. 5. create a sip user account on new customer and route his calls to trunk of termination carrier. test and make sure all calls connect properly. give me a walk thru of the steps taken to do each ,.
Dear, All Viewers we are looking for someone who can help us to create Asterisk Native Dialplan Using ODBC to control callflow according to our logic without using any AGI, just using asterisk pure dialplan. Our goal to make lightweight fast execute & 0 resource using, please only experienced people bid, waiting for your bids, Thanks for your attention.
We need Push Notification feature via Event Based Routing in OPENSIP. Please BID if you have knowledge of OPENSIP.
Hello I want to make Asterisk server for Dynamic Black list on Destination numbers As I have calls running on my softswitch I want to add this FreePBX server to block the calls under the following criteria: 1- if the Destination number hit 5 times and total duration less than or = 10 Sec 2- if Destination number make 5 hits in less than 10 minutes 3- if Destination number make 10 hits in less than 12 hours If one of the above then add this destination number to Blacklist Whitelist criteria: 1- if number is answered with Duration more than or = 40 sec add to whitelist So I will be sending calls to this Server to filter then send it back after filtering if Blacklist number = reject call if white list number = send back the call with prefix 777 if number is new and not yet in blacklist or whitelist = send back with prefix 888 I hope the Idea is clear enough, If any thing to ask you can massage me
viciial server setup and installation on centos with configuration for calling . need to start today and assist in installation and configuration. Thanks
We have a cloud based phone system running FusionPBX. We are starting to have a large number of phone / system orders, and need to setup provisioning correctly - so we can drop ship handsets to a customer site, and then the correct details get provisioned to the phone. We need to build provisioning templates for our standard phones (Gigaset N300 / N500) and a variety of handsets (S650H, CX430H etc.) We also need SNOM, and LG / Nortel IP8840 as well. We would also like this functionality for our broadband routers as well.
We need create gate call forwarding from SIP to some messengers Viber, Skype, Whatsapp, Telegram if it possible. And send messages to these messengers. ------ Необходимо создать гейт SIP в мессенджеры Viber, Skype, Whatsapp, Telegram (в какие возможно). А так же отправлять в эти мессенджеры (плюс facebook messenger) сообщения.
We need a specialized consultant to install a FreeSwitch server in the Microsoft Azure cloud with the following capabilities: 1. Support for video conferencing with Vertor Communicator; 2. Support for secure communication with TLS and RTPS; 3. Communication to SIP Trunk using [url removed, login to view] as provider; 4. Graphical interface for creating extensions; 5. Possibility of using database for creation of extensions; 6. Installation of Portuguese language for all functionalities. The project should be built together with our technicians to be trained during the implementation of the environment.
Leave a recorded voicemail without ringing the phone, to a list of numbers that i can load. TEXT FILE CSV upload wave /mp3 file to be sent Voip or GSM mobile phone to pc You must have previous experience on ringless voicemail drop Please present your previous work you have done only get paid after delivering the job