Asterisk PBX, like any other PBX, is a complicated subject that is best handled by experts. If you are a pro in this field, then you should bid on the many jobs at Freelancer.com.
Asterisk PBX (private branch exchange) is implementation software. Created by Mark Spencer in 1999, the software simply allows connected telephones to make calls to each other and also to connect to other services. The name is based on the symbol asterisk, (*). For Asterisk PBX to function as it should, the configurations must be on point, which is why this should be done by an expert.
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Hi guys We have have PBX call center UcSaaS and we are looking for developer to build customer support module to integrate inside PBX Tickets-Deals-Chats-email Source chaneel (email-facebook messanger-whatsapp) so each incoming calls or messages from social messager will create contact and ticket auto or manually tickets To save time we need like customer support module of workflow and design Full API is available from PBX
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
Hi guys We have have PBX call center Ucaas and we are looking for expert developer to build mini crm and integrate inside PBX to make interactive calls Tickets-Deals-Chats-email Source chaneel (email-facebook messanger-whatsapp) so each incoming calls or messages from social messager will open auto or manually tickets To save time we need like workflow Full API is available from PBX
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:220.127.116.11 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
Send sms when called a Twilio number, send a link to continue in Whatsapp or sms, answer sms incoming messages as conversation on web interface, have login for customer, agents per customer and supervisor. Only agents can see active sms conversations, closed conversations are for reference and can be seen if new conversation is active with contact number, provide rports for daily, weekly, monthly conversations and time.
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