Need someone to install, configure and test asterisk a2billing (for calling card service use) on a Google Cloud VM. Only bidders with proof of prior/similar work will be considered. Asterisk realtime experience a plus. Thanks.
1- We want my VOIP app to 1- Keep Awake and 2- Push Notification always on even app is closed in opensips for android first. Must be familiar with linphone, opensips and firebase. 2- Upload to google play. 3- Give me source code after finish the job.
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A2billing with asterisk installation Script + Configure the basic calling card and DID function
...softswitch If you want to Use asterisk Use JAVA-AGI database must be PostgreSQL Priority musbe be have to work you can use any good Engine like asterisk,FreeSWITCH,Kamailio,OpenSIPS,etc More info check Attach file here is all screen shoot i need a complete wholesale Softswitch with a web management panel If you can make it please bid i dont have time
...and Kamailio system 2) The engineer must have an in-depth knowledge of SIP layers 3) The engineer must be able to understand and troubleshoot systems such as FreePBX and A2billing instantly 4) The engineer must be able to create a good security on the system 5) The engineer must have good web development knowledge for creating an online system with
1- We want my VOIP app to 1- Keep Awake and 2- Push Notification always on even app is closed in opensips for android first. Must be familiar with linphone, opensips and firebase. 2- Upload to google play.
1- We want to use 1- Keep Awake and 2- Push Notification always on even app is closed in opensips for android first. Must be familiar with linphone, opensips and firebase. 2- Upload to google play.
We want to use 1- Keep Awake and 2- Push Notification always on even app is closed in opensips for android first. Must be familiar with linphone, opensips and firebase.
We have A2billing server and we are building an app for our resellers to topup customers through app. We need the working API to Create and Manage customers - mainly Create customer with Callerid (including sip account) and topup.
...Asterisk, FreePBX and A2Billing Server installed in Google Cloud on a Free PBX. I need help to configured it so that I can bill departments in my I have already installed A2Billing asterisk server and set up a Sip Trunk to my VoIP provider. My internal free PBX extensions are working, however, I need help in configuring the A2Billing servers. I have
Server OS: Centos 7 Software: Asterisk 13 + FreePBX + a2billinng Network: Dynamic IP behind NAT Problem: - Call receiver no ring - No audio as it should to tell balance, dial 9 for sip or iax Because this desktop behind NAT, i'll provide Teamviewer or Anydesk access
I need help in setting up an OpenSIPS server and creating a SIP Proxy that alters some headers. I have knowledge of VoIP and SIP, but no experience in OpenSIPS. I'm a professional developer and do systems administration so i should be able to learn quickly. I would like someone who can teach me the basics, and prove me with answers to questions i
Hello there. I am looking for an experienced VOIP Website developer. Please only bid if you have past experien...there. I am looking for an experienced VOIP Website developer. Please only bid if you have past experience in VOIP Web development and knowledge of DID Provider integration, A2Billing etc. More details will be discussed in chat. Thank you
I noticed your profile and would like to offer you my project. Do you have experience with GoIP VoIP gateway, and A2Billing? I would like to deveop a system similar to [login to view URL], but with different business model. So, it should efficiently setup and operate the GoIP (and similar) equipment to improve VoIP GSM termination. Thanks, Marcelo
...connect a Client to a Carrier that is scalable. However, we do not want to disclose any of the Carrier IPs to the Client and Vice Versa. I've done some research and using OpenSIPS and RTPproxy can help with this but I am having trouble setting this up on AWS. For the proof of concept, we can use 2 PABXs one to act as a Carrier, another to Act as a Client
...someone to complete the initial configuration of an audiocodes mediant VE box ASAP. Initial Configuration is for 2 x IP-IP connections (Upstream Providers) and 1 x PABX / Opensips / downstream. Initial network configuration is completed. Configuration is required for the above + basic call routing and SIP headers. with the requirement for a basic configuration
Can you make Linphone wake up with flexisip or opensips or freeswitch push notification and receive incoming calls or sms ? Freeswitch and linphone is already up and running my budget is $500
Hi I would like to configure Kamailio or OpenSips for load balancing of some freeswitch servers and I would like to use ASTPP as billing for that system. I would like to have about 1500cc Thank you
I need to have 2 servers. 1) FreePBX as a PBX and 2) A2billing for Billing system. I'll supply both servers. Just need the Configuration and interoperability with A2billing.
...allowing them to top op their accounts for DID usage and termination services. 5. Use Sip Router (Kamailio, Openser, Opensips) Load Balancing, Registrar, Routing outgoing calls by LCR for termination 6. Integrate a billing system (A2Billing or suggest other). If A2 Billing, code for email dunning or ability to download counts receivable (late payments)
Hi Pragati S., I noticed your profile and would like to offer you my project. Do you have experience with GoIP VoIP gateway, and A2Billing? I would like to deveop a system similar to [login to view URL], but with different business model. Thanks, Marcelo
Looking for Android developer to modify LinPhone app and integrate couple of features. Please apply if you worked with Li...Android developer to modify LinPhone app and integrate couple of features. Please apply if you worked with LinPhone before. Must have experience working with Asterisk, A2Billing Review the attached file for more details.
Simple SIP redirect Routing system ,listen on port 5060 .It authenticate the incoming SIP invite source IP address , retrieve data by connecting to Mysql or Sql server, get values that should be appended to sip contact header edit/append sip contact : content ,then send it back to source of invite ,wait for ACK and send invite again if no ACK within 3 seconds (max retry 5 ). 2-Installat...
Project is to configure a2billing act as billing platform for go-autodial/vicidial with all calls to go-autodial/vicidial setup to be authenticated, rated and charged via a2blling before being routed to the go-autodial/vicidial call center. Documentation of implementation steps expected. A2billing and go-autodial/vicidial installed on separate vps's
OpenSIPS/Sippy Carrier Routing - Stage 1 either an OpenSIPS or Sippy box will act as an outbound traffic router for multiple carriers. Calls will be sent from already running Asterisk boxes to the OpenSIPS/Sippy/SippyGO box (we will prepare base OS, OpenSIPs/SIPPY core installs) Media needs to be proxied with RTPProxy/etc through the OpenSIPS/Sippy/SippyGO