...to configure FusionPBX with PowerPBX hosting. [login to view URL], [login to view URL], Debian 8. FusionPBX system already installed. Configure Dial Plans, PBX Extensions, I/O Trunks, Voicemail, DISA, IVR & local number display on the go, Call Forwarding, Call waiting, Linphone setup and instructions, call recording, setup sip trunks. for 4 - 6 lines. PHP or LUA
...Attendant Voicemail setup. Right now I have 2 voicemails when someone calls. Day/Night Menu and a Que announcement Voicemail. When someone presses #10 they will go to this announcement. I dial *91200# to get to this recording but i don't know how to link it together. So for example: The Customer calls the phone number and they hear main voicemail " hello
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
...support. Responsibilities: • Greet and assist clients and visitors as they arrive to the office. • Handle all incoming calls and appropriately transfer to attorneys, staff, or voicemail. • Ensure all mail at reception is delivered to the appropriate individual. • Manage multiple conference rooms, set-up for meetings, videoconferencing and teleconferencing
...****************************************** GENERAL VOICEMAIL We're sorry we missed your call we appreciate your business and will get back to you before the end of the business day. ************************************************************************************************** CUSTOM VOICEMAIL Thank you for calling the Custom department, please
I have a telephone exchange model Nec Sl1000, i need someone that is able to change one setting (the voicemail time in the setting that we are able to get from browser IP) May the setting module is this (www) .[login to view URL] (.html) If you u have to search I need a reply like this . Open this setting . Change this field
...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
Task #1 We need to monitor & intercept incoming calls and be able to play a recording that allows the user to press 1 or 2 and do a specific task (leave voicemail, text caller, allow caller through to ring recipient, etc). Task #2 Detect outbound phone calls. Both tasks need to be done on Android & Apple phones. Open to any strategy that can get
...preferably were constructed to work in version 11 and 13. The ideal candidate will be able to stand up their own Asterisk development environment to build and test in. An asterisk module which will allow us to, while placing an outbound call and playing a set of sound files, simultaneously listen for a single frequency tone (answer machine tone) and or DTMF
Looking for an expert to configuration FreeSwitch installation with ZRTP enabling. Need someone who is highly skilled around FreeSwitch, VoIP and Linux administration.
Hi franciscogzz, I noticed your profile and would like to offer you my project. We can discuss any details over chat. I am interested in building a ringless voicemail through twilio. I had to put something in pricing below pls ignore it.
...have voicemail where someone didn't hang up and it recorded them on another call. You can hear the person that called me ok, but the person on the other phone is difficult to make out. I want the audio cleaned up so I can understand both parties better if possible, especially the person on the other phone (not the caller that left the voicemail).
Require an audio file (mp3) for a voicemail message with the following script clearly and concisely spoken in a female neutral English accent. "Thank you for calling Better World Gadgets. We are super happy to help you! Our normal business hours are 9am-5pm Eastern Standard Time. We are currently experiencing a larger than expected call volume and
I have a Cisco Unified Communications 560 for Small Business (UC560) that is causing us problems, We need to setup the voicemail when nobody picks up the phone so the corresponding department can retrieve the messages addressed to [login to view URL] other important issue is that the phone system sometimes works extremely slow. There are a few bits and ends that
...and his interests. 5. The main screen allows the user to display people according to their relationship with him. 6. The chat will be in writing, sending photos, sending voicemail and calling (the personal profile includes the choice between receiving calls, voice, and photos) 7. The user adds his favorite friend so that he can email him if he is online
I need software that will allow me to download an mp3 file send it to any voicemail of any mobile phone service. I need to be able to put my google drive number in as the message sender number. I need it to work with my mac (15 inch late 2011). [login to view URL]
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
ASTPP 3.6 running with freeswitch i want to listen live calls
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
Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload
...budget don't waste your time on chat if you bid less and then increase it after dont bid Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls
We are currently using a provider for this service and would like to use our own system to leave voicemails on cell/mobile phones without the phone r...and would like to use our own system to leave voicemails on cell/mobile phones without the phone ringing. Please do not apply if you have not already worked on a Ringless Voicemail development project.
Need ANSI C Programmer with FreeSWITCH module programming experience. Also, experience with AF_UNIX named sockets. Module needs to buffer sound bytes of audio from A leg into memory, package and deliver to another process upon silence threshold being met, and continue until timeout or message received from socket. You could start with mod_dptools
We are looking full stack developers to build a telephony system based on engine Asterisk or Freeswitch. We need to build the whole system including a client area and manager interface. Experience on VoIP/Telephony proyect is a must. Please contact us for more details.
I have run into a roadblock and am looking for an expert to help me extract a wav file from an email inbox that is sent from our voicemail system. This should be a very brief project once the function is built we will take over and complete implementation. This requires a knowledge of PHP and IMAP functions, specifically reading from an exchange mailbox
...content must have the skill’s definition, advantage of the skill, where the skill is applied, why the skill is important in a business, etc LiveCode Attorney Flask iMovie FreeSwitch Google Cloud Storage Typescript Adobe Illustrator Scrapy Rust Qualtrics Survey Platform Snapchat Weebly Swing (Java) CSS3 Instagram API Brand Marketing Material Coating
Hello. I am looking for someone that can build a system that will deliver voicemail messages to external numbers without ringing the phone number. The system I am using is FreePBX v14 with trunks from Twilio. This is also knowen as VoiceCasting. It's important that you understand that the phone numbers are NOT phone numbers or extensions within the
...Internet. Compared to traditional phone ISP’s, VoIP has many benefits, including lower costs and increased functionality. More specifically, you can enjoy using a customized voicemail and attendant menus, call forwarding, on-hold messages or music, using your cell phone when you’re out of the office, and much more. VoIP is the future of business calls,
Im setting up a network for VoIP. Am looking for an engineer that can assist in setting up the network with Opensip as the SBC and loadbalancing to 2or more Freeswitch servers. Security and firewall is a must in the network. Must have the ability to design the network. All equipment will be in the datacentre.