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
[updated job: no bare minimum requirement, no source code build] I need a FreeSWITCH (FS) configuration for the following functionality: 1. Interconnect internal endpoints 2. Link with an existing Asterisk PBX 3. Link with an external SIP trunk for incoming and outgoing calls 4. Configure basic CDR, voicemail, IVR and conference 5. Configure
...with Asterisk Voip experienced. Min. 5 years hands on experience. Mandatory skills: Centos7,6 Asterisk Optional Skills Expeience with 5 Softswitch Asterisk, Freeswitch, Kamailio,Opensips,vicidial,A2billing, Free pbx, elastix, PRI & PSTN card installation, call centre setup, Advanced IVR, Vmwire Esx, web-meetme ,Cloud solution using asterisk
...requirements/functionality: Translating random numbers to real names in the database and kazoo Knowledge of Kazoo, Erlang and CouchDB required. Specific technologies required: Kazoo, Freeswitch, kamailio, Erlang, Couchdb OS requirements: Linux Extra notes: I have have just set up a VOIP server using Kazoo from the 2600hz which I intend to use for commercial VOIP
I need a FreeSWITCH (FS) configuration for the following functionality. It should connect: * 2 Linphone endpoints, extensions 100 and 101 * 1 Linphone endpoint + 1 Yealink T41S phone on the same extension 102 (both should ring simultaneously on incoming call) * 1 external SIP trunk provider for incoming and outgoing calls For outgoing calls
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
This project is for install ASTPP Billing with Freeswitch in a DigitalOcean Droplet i need the last version installed in Debian 8 please just text me if you have worked with freeswitch or astpp before thanks
1- To configure the open source Linphone Flexisip to interact with my voip server using TLS protocol. 2- Customize the app to interact with my freeswitch/asterisk server to do registration , logo, color, configuration, etc customization - show me how to do this
I require a price for experienced programmer to install ASTPP billing platform with Freeswitch onto a remote server. Experience will be essential.
I have an issue, with transferring between queues, I'd prefer att_xfer but be aware the issue occurs with uuid_transfer as well. So fixing either should fix my problem. I need you to be able to accomplish this in your lab before I will award you the project, additional work to follow, I'm looking for a long term support resource and this will be the test. Here is what I'm doin...
I need someone to help configure gateway on freeswitch for my ict dialer. Very straight forward task. Thank you.
Setup of FreeSwitch to correctly handle Refer with Replaces - Only for experienced SIP literate developers - I would like to have FS be able to process the following and create an invite to process transferring the call - REFER-TO: <sip:[url removed, login to view];transport=Tcp;maddr=[url removed, login to view];ms-opaque=1befc8e31af6a72f?REPLACES=d71cf52b-2961-4
I am looking for a freelancer to help me with my project. The skills required are Asterisk PBX, Cisco, FreeSwitch and VoIP. I am happy to pay a fixed priced and my budget is $250 - $750 USD. I have not provided a detailed description and have not uploaded any files.
We need help to configure custom SIP gateway in our FusionPBX for the outbound calling service. Currently our SIP provider unable to provide us direct SIP settings to configure them with our FusionPBX as they only providing us Mobile Dialer based SIP client which we are unable to configure as SIP gateway. We want to configure that Mobile SIP Dialer (SIP Settings) with our FusionPBX Gateway Pro...