...and set it as default, so just need to initiate a call on gsm network using core android functions and audio will be automatically forwarded. Codecs: Minimum g711, desired g729. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please let me know before we make a deal. Summary
I want use features of freeswitch on Astpp (for default, we cannot use the dialplan files and we don’t have the features how transfer, queue, etc on Astpp). You can make the features on dynamic dialplan or use dynamic ans static dialplan simultaneously.
Hi. I want the my Astpp server use dynamic and static dialplan symultaneos. I want use static dialplan for pbx functions (transfer call, IVR, etc) and dynamic for make calls.
Hi, I want my freeswitch with ASTPP has a transfer function when I dial a *2 and *1. Example: I receive a call. I type *2 Prompt say: Type the extension or queue to transfer I type the extension or queue number The system transfer the call (I am on line) I call to another extension or queue I drop the call The caller are transfered to extension or
==================================================== Please, I want someone who already have experience with Csi...PBX - Enter your PBX Name Username - Enter your Username Password - Enter your password 3. Call logs will be only numbers - not the whole sip URI 4. In settings, G711a,G711u, G729 will be enabled by default 5. Help : Link to our web site
Hi, I use PBX FusionPBX and Freeswitch, everything working ok. but problem is when app in phone exit, app can not wakeup to receive call signal. I want integrate push message to ios/android device to wake up the app and receive call
Setup Session Border Controller Kamailio with Fusion/FreeSwitch Hosted PBX Requirements: • Kamailio SBC experience and skills when working with PBX • Understanding of Fusion/FreeSwitch PBX working with Kamailio • Understanding of SIP registrations and load balancing via Soft Clients
Hello there! We are currently facing a problem with our PBX system, Fusion PBX, which is a GUI for Freeswitch. It is similar to 3CX or AsteriskPBX. The problem we are facing is that users can not click on
...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
help me fill out a mock t2 and t1 return given income earned from capital gains, eligible and ineligible dividends inside RRSP/IPP acct and private corporation. Please note you are going to fill out a CANADIAN T2 and T1 so please cdn tax trained applicants only - thanks.
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 work with complex radio signals in Intel IPP, I store and I carry out over them operations in the Ipp32fc format (complex float). In an investment example of a complex radio signal of [kirjaudu nähdäksesi URL:n] (first column Real, the second column imag). I need to build a signal in degree which can reach great values, up to 128 (complexSignal^128). I do
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
...PV: Sectors, Technology and Ecosystem Solar PV: Business Opportunities and profitability matrix in India Solar sector: Role and responsibilities of stakeholders in India Be an IPP: Role, Returns and Strategy for Solar EPC Business in Solar: How to manage and make it profitable O&M/Assets Management Business: Model, Procedure, Profitability and Resources
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
I need my "Impact Payment Partners" Logo converted to a vector file format. And then I also need to have the "Impact Payment Partners" logo to be edited to say "Impact Pays" and I need this in the vector file format as well. I have this logo in .tiff form but it will not let me upload it. Let me know if these files do not work.
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
Hi Kym, Thank you for your help last month transferring our IPP Guidelines document into a new template. I really appreciated the high standard of your work, and your responsiveness. I have a further similar piece of work I'd like to request your help with if you are available. We have now published four of the updated chapters in draft form on
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