Here are some essential steps that you should not overlook when preparing to launch a new business from your home.
he desarrollado una ivr en issabel. tengo todo el sistema de marcado gestionado con php, el problema es que cuando se ejecuta el bash para para marcar las llamadas salen pero no conectan a la ivr, nesecito soporte en este aspecto, estoy buscando especialistas en asterisk y issabel que hablen español, adjunte la consola de click y el bash que desarrolle
We need a Technical person who can install a functional VICI Dialer on a server of his choice and knows how to configure softphones like Zoipher, eyebeam to the Dialer for US Calling. He will be paid for Installation as well as maintenance of the dialer very month. He must know about AC-CID and the Outgoing call must be that of the Customer. Must have expert knowledge of Installing VICI Dialer and Configaration
WebRTC SIP (Move from to JSSIP) Using Typescript ... i duplicated issue on ctxsip so i think jssip might be a better package as its maintained.
i have asterisk sip server and its working fine with any RTP i set on my system interface. but in same VM if i copy and make a new server when i try to set any new RTP its not working i dont know whats the issue audio is gone. so i need you to any how create a system where i can set any rtp as i want. like or or anything as i want. (we dont use our public IP as RTP we use any IP on our RTP like a eth0:1 interface ip is:126.96.36.199 so i will use this as a RTP) you can use any sip server or anything as you want. i just want to use RTP thats it. you can to setup this your local system or if you want i can give you server dont ask me any payment before test. if you can show me its working and audio is fine you will get payment with bonus.
I have freepbx already installed and goip4 gateway already installed and configured. I want to configur freepbx to connect with goip4 gateway 3 lines (simcards) 3 SIP user : user 600 recive call and working with line 1 from number start with 06 user 500 recive call working with line 2 from number start with 05 user 700 recive call working with line 3 from number strat with 07 variables if user 600 dont respond he redirected to voicecall to leave a message 1 after that send sms offer 1 if user 500 dont respond he redirected to voicecall to leave a message 2 after that send sms offer 2 if user 700 dont respond he redirected to voicecall to leave a message 3 after that send sms offer 3 ps dont change any network config on goip4 gateway.
I have already did the full process from verifying Facebook and verifying Twilio with WhatsApp with my own number and I have verified the template But I have an issue and I need someone to help me with it Error 11200
Hi guys, I need create and setup a IVR campaing in our Vicidial Server, to sent IVR to ours customers, and finally download a complete report with the calls status (answers, failed, no response, etc)
mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Dialplan manipulation etc. Needs the ability to work Eastern Standard Time hours for supporting us and our clients
Hello, We are looking Kamailio and Opensips Expert to integrate the below Kamailio and opensips module We need a return Invite as per the below URL configuration Can you please help us Thank You
We have GSM GW Like Dinstar, Ejoin , SK. GW have Sims. Each Sims have own number So we have to Dial number local so need to develop some solution where GW connect some dialler and with help of dialler we dial specific number from GW sims.
There is a campaign for incoming calls, the calls that enter the agents are recorded, but the calls that enter directly to an extension registered in a softphone are not recorded, these calls are required to be recorded and can be seen from the dashboard
Hi, We are having IPPhone System DVX-2005F linked with 2 Lines and 3 SIM card Lines . we are having issues with : - it takes one extra second for the caller to get in the central. - we sometimes can not make calls to mobile numbers and 9200 we need an expert VOIP with Good D-link experience
I need configure integration chat with clients website to Goautodial, and integrate button ClicktoDial with clients website. The idea is that on the client's page there is a chat and a call me button, so that it is integrated with a goautodial campaign and can be managed by the advisors.
The istallation of the softswitch will be used to set up a VoIP platform for a voip retail business start. The softswitch will be installed on a cloud server, and will be configured ready to start using and intergration with an app. The last step will be testing it with an app.