Hi I would like to configure Kamailio or OpenSips for load balancing of some freeswitch servers and I would like to use ASTPP as billing for that system. I would like to have about 1500cc Thank you
Hi. I am new on ASTPP. I use the link([login to view URL]) to manual install the ASTPP. I configure the trunks, rates, sip, gateways, etc. All are configured. When I try to make a call, show me error on fs_cli and the call hangup: 2018-01-13 13:00:25.568487 [ERR] switch_odbc.c:368 STATE:
I have a fresh installation of ASTPP , i need the following configured: 1. create origination carrier ( customer) 2. create termination provider [login to view URL] 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
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We need a billing and LCR solution for FusionPBX solution, Probably ASTPP, assistance required to guide configuration
I currently have a VoIP Mobile Dialer (iOS & Android) for Asterisk A2billing, I need to slightly modify to work with FreeSwitch ASTPP Billing. Please let me know if you're available for this project. Thanks,
We need to install the latest stable release of OPENSIPS and ASTPP in the server and then establish calls between users using SIPML5. Calls should be established between - Chrome to Chrome browsers - Chrome to Android devices having the app installed and vice-versa - Chrome to IOS devices having the app installed and vice-versa - also
Hello guys, This project is belong to who make are expert. i need to install FreeSwitch on my server and install ASTTP on that. then doing this configs: 1- Activate G729 Codec on that 2- Change transfer protocol to UDP 3- activate just SIP protocol on server. (disable another protocols) 4- Enable Video Call feature on freeswich if you not robot please add "Swh" in your bid...