...Asterisk must not bridge the final connected call. Asterisk must remove itself from the call-flow once the call has been transferred. The other pbx is currently setup under SIP and PJSIP with the same remote ip address. We are using the Asterisk pbx as an auto-attendant only. Our main pbx will send calls to the asterisk pbx and expects the calls
Hi, You previously worked on my FreePBX system. I would like to do a few things and could use some help. - Want to setup an extension that sends the calls to a SIP URL. I know this can be done in Asterisk but not sure how to do it in FreePBX - Perhaps it is time to upgrade my version, I am very behind. Largely things are working though. I am running
Need someone who has a good understanding of SIP architecture. Task to be done: 1. Work on outlining the architecture documents. 2. Setup SIP server like flexisip, Astrix. 3. Support our developers in integrating softphone in iOS and android app. 4. Support our developers in integrating SIP client in hardware embedded devices.
We are a VoIP provider selling US-48 Termination at $.0059/minute billed in 6-second increments. We offer a very competitive lifetime commission for every client you bring. Ideally, you're able to support these clients from a technical perspective. We also have a few clients that we need to supply support for. This includes hosting, debugging
i want to build a custom crm with multiple features, please contact for more details.
I have voip program. I need assistance in helping to set it up and to successfully run it. I need someone who can work quickly in making a build/app from the compiled project, renew outdated modules if needed and to set me up with a fully working app and to get me up to speed with everything what I need. Further expansion goals are possible.
I need someone who could develop quickly a basic voip program who will have additional voice packet encryption. You can take any open source program without it and set it up for me and tweak it. This should be done on Android platform. I will prefer someone who already had worked on it and preferably someone who understands how packet security works
Hello, I'm looking for a freelancer who knows Asterisk well and can install it on a new VPS for IVR/SIP/VOIP calls after the installation, we'll set the SIP and try to call and recive phone calls * the VPS is in europe * I can install on the VPS one of the most common OS like Centos and ubunto * I rather to install DirectAdmin on that server
stand alone click2dial, webrtc2sip a sa service, 1- server will not use outside resource 2- very stable call button is m...webrtc2sip a sa service, 1- server will not use outside resource 2- very stable call button is must (secured from third party) 3- new account will get all needed actions (new sip account and html code to paste their web page)
...be completed and I will share the Dropbox link and how to run it (I used OMNet 4.6 running in windows 7 32bit). in the project I used OMNeT++ as network simulator to build 2 VoIP networks wired and wireless one.(done) by using 3 different codec like G.711 (I only used G.726) and 3 encryption algorithms (RC4, AES, and Blowfish)(done need some improvement)
On the 28th of April I would like to have a paint and sip party on the south side of Chicago. Attendants will be mostly couples ages ranging from 25 to 40, Black and Latin Americans, and family members. I will purchase all items necessary. Just need a painter.
...improve the features concerning CTI,API,mobile integration, conferencing etc. Integration with Office 365- Skype for Business (calls in/out via fusion pbx trunks) CTI with instant messaging for freeswitch-Fusionpbx with extensions/status, LDAP, instant messaging, call logs. The cti need to be improved with SIP videocall features and conferencing up
User Story: When my new JS SIP client running in Google Chrome Browser is receiving a call (routed through our Asterisk PBX) I want a HTML popup in Salesforce with 10 key customer information showing up. Contact Name + 2 more contact fields Account Name Language Won / Open Opportunities We run our own servers such as Astersisk, FreePBX on AWS