We are developing a telephony to chatbot tool that uses the open source Voximal Suite [войдите, чтобы посмотреть URL] with our Pandorabots chatbot https://home.pandorabots....take over the phone calls with the customers. In that regard we are seeking assistance in integrating a capability where the customer service agent can take over the call via SIP (VOIP).
We have a few Skype Connect profiles with inbound numbers at...script should (persistent) connect to Skype servers, login and answer the calls by returning a recording (depending on the inbound number). That's it folks:-) It's all standard SIP, pretty straight forward. --> $100 FIXED (it's max 3 hours work, MAX, 30 minutes if you've done it before)
hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [войдите, чтобы посмотреть URL] i want to run this program on a router Router Link [войдите, чтобы посмотреть URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580
Help programming sip endpoints, applications voice & sms in [войдите, чтобы посмотреть URL] I need to connect my [войдите, чтобы посмотреть URL] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [войдите, чтобы посмотреть URL] Prior experience
In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number
Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this
i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole