...installation that we want your help with transitioning if we the good times what we see on FusionPBX. I will want you to show me at least a few screens of a FusionPBX (or FreeSwitch) environment you manage so I know what you're talking about. this first set up is essential because it forms the basis for what we do in the second project. I would be looking
... SoftSWITCH - We are interested in welcoming a person with a profound interest in Telecommunication SoftSWITCH, Oh! [Nice to Have] knowledge in leveraging Kamailio with FreeSWITCH or Kamailio use of Asterisk latest version as a media gateway. The SoftSWITCH with SIP proxy server, which to provide SIP client registration and call routing. Processing
...traditional GMS getaway) This mobile application can work with any Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Windows server based VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server
I need a way to do fax broadcasting via fax over IP (T38). Therefore, I need someone to put together either an Asterisk server or a Freeswitch server which can broadcast faxes using whatever existing fax broadcast software that would work best with the server. I have an account with [url removed, login to view] which I'd like to use but I am open to
Hi Aqs Y., how are you? hope you are doing well. i am looking for answering machine detection for FreeSwitch to integrate with the IVR on auto dialer and Ringless VoiceMail solution. looking forward to hear from you. Thanks
Hi stwrtmk, how are you? hope you are doing well. i am looking for answering machine detection for FreeSwitch to integrate with the IVR on auto dialer and Ringless VoiceMail solution. looking forward to hear from you. Thanks
I am looking for answering machine detection for FreeSwitch to integrate with the IVR on auto dialer and Ringless VoiceMail solution. looking forward to hear from you. Thanks
Salam Bilal A., how are you? hope you are doing well. i am looking for answering machine detection for FreeSwitch to integrate with the IVR on auto dialer and Ringless VoiceMail solution. looking forward to hear from you. Thanks
I need to compile the latest v...on completion. you need to setup the VM and install the latest android NDK, SDK and JDK. All the required packages must be installed by the selected candidate. We need the G729 and SILK codecs on linphone, selected candidate must, install android studio 3.4 on the VM and allow the application to run on the emulator.
We need an application that runs on Android phones (4.0 or above) and can receive calls through SIP and forward it to the GSM n...IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Be able to store a CSV with all calls made
Hi voipli...IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: Android, VoIP
we want to get developed a Class 5 Voip Soft switch sports sim and G729 Codec. Class 5 SoftSwitch development renders intelligent programmable routing that includes a management interface for configuration and reporting. Clients can log on to the system to manage calls; they can access IP gateways and get exclusive AAA billing solutions for telecom
hi i want to deploy Freeswitch with one of the existing opensource Web administration interface and configure only 4 Extensions budget is 50$ (please do not bid initial and do not bid above this)
...with android and iOS in our logo and company information, and integrate the apps with our existing ASTPP PBX server , required features are listed as follows 1. audio calls (g729, ulaw, alaw, Opus, iLBC, GSM) 2. video calls with video preview (H.264, H.265 and VP8 codecs supported) 3. Multiple calls management 4. Call transfer, pause and resume
Hi all, We need an expert FreeSwitch developer solution for the Android call app with which- 1) Call is working fine and but when mobile switched off and on then call isn't connecting remote user. 2) If app is in background for hours then call isn't connecting for remote user. Please bid only if you have done something similar. Thank you and have
We have an codec error of FusionPBX Codec Passthrough issue mod_g729.c:102 This codec is only usable in passthrough mode! 2019-03-26 07:37:44.698244 [ERR] switch_core_io.c:1434 Codec G.729 encoder error
I'm new to FusionPBX/FreeSwitch. Can't set up the same thing I've set up in minutes in Asterisk. Need remote assistance/general consulting or just a setup example in accordance with my infrastructure conditions. My FusionPBX is a VM that has only one LAN NIC. It uses pfSense VM as a gateway. pfSense VM has both LAN and WAN links with a public IP on
I am looking for an experienced FreeSwitch engineer who can work with latest freeswitch version with SIP Trunks to do following tasks I will be passing destination number , CLI , ring duration , max call duration , audio file , unique call id to Freeswitch (JSON API) , the dial plan should start dialing numbers and plays a voice file when the call
I want someone expert in ASTERISK and freeswitch and ios to send calls over bluetooth but to use whatsapp or viber instead of gsm dialing to be smart enough to tell asterisk use whatsapp when dialing (send calls direct to whatsapp) android or iphone whatever they can fix
...attach more huawei usb modems to the server if needed. USB hubs will be used to attach the usb modems. - The server shall be able to terminate voip calls comming in codecs g729 and g723. - Other standards codecs like G722, G726, alow, ulow... shall be supported - Send ussd and ivr to each sim card to topup and credit check. - Display how much credit
...server Server B = Asterisk Client server Explanation of the scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quantum gateway
We are new small VOIP Services provider and We need a SIP Voip Softphone App for mobiles mainly, eventually for windows too. The App would be similar to Zoiper, Bria, SIPGo... SIPGo, SessionTalk, etc. but customized with our Logo, our proxy domain and most important the capability to use all video and audio codecs including the now free licensed G729.
...in right context, it is desirable, but not necessary, with experience and knowledge within • On premises deliveries of system and installation script • Telephony and SIP (FreeSwitch, Swyx (a German PABX) and eventually Asterisk) • Record keeping, eventually through a PHP framework • Access rights for different users, we work with an access token today
Voip expert with PBX (asterisk/freeswitch) and dialers (vicidial) experience to help deploying a special use case related to a voice bot product (not for human agents) A relationship with our low level developers. A long term job relationship is promised to whoever will fit the job
...your profile and would like to offer you my project: We have setup a fusionPBX/freeswitch server and are using sipjs as webrtc clients. Everything works fine except once scenario: extension A calls B, B picks up. So far everything is good. Now A hangs up, but freeswitch never signals the hangup to B, so B still seems to be on the phone while A is long
...accomplished on Freeswitch with mod_avmd. Perhaps it can be ported to Asterisk. [kirjaudu nähdäksesi URL:n] We need to have some dial plan variables for this module that allow us to adjust the tone duration and tolerances for amplitude and frequency. If one references the Freeswitch module
Looking for an expert to configuration FreeSwitch installation with ZRTP enabling. Need someone who is highly skilled around FreeSwitch, VoIP and Linux administration.
We use FreeSWITCH and opensips now. We want to redevelop a GUI for our Telecom System. If you are expert in this field, please happy to bid, We are looking for only experts now, Thanks.
We have setup a freeswitch / fusionPBX server and are using sipjs as webrtc clients. What we cant figure out is how to make presence work. We subscribe on the server, get an ok back, but we never get a notify with status info about the peers.
Hi Bilal A., I noticed your profile and would like to offer you my project: We have setup a freeswitch server and are using sipjs as webrtc clients. What we cant figure out is how to make presence work. We register on the server but we never get a notify with status info about the peers.
Client wishes to write a next generation layer on top of current free switch layer. The technology details are :Python, R, knowledge of free switch. (Very good coding skills) Experience level :2 to 3 years. Contract Duration : 6 months to 1 year. Start date : 1st March. Location : Pune Only locals please
Client wishes to write a next generation layer on top of current free switch layer. The technology details are :Python, R, knowledge of free switch. (Very good coding skills) Experience level :2 to 3 years. Contract Duration : 6 months to 1 year. Start date : 1st March. Location : Pune. Only local people please
Hi altr, I'm looking for someone to help me resolve an issue i'm facing on freeswitch. I'm using freeswitch for wholesale scenario, When i receive 181 call is being forward sip message from supplier, it is being absorbed by freeswitch and not sent to Leg A. I don't want to spend time to figure it out, i just need you to tell me how i can make sure
...faça ligações através de servidores PABX's. O principal ponto do projeto é que o webphone deve efetuar ligações através de vários servidores PABX do mercado, ex:. Asterisk, FreeSwitch, intelbras e etc... Caso seja necessário uso de WebRTC Servers, fica a escolha do desenvolvedor. "Podemos analisar outra lingu...