Asterisk Jobs

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Fixed Price Budget - Expert ($$$) - $150 to $250 - Posted
We're looking to have a knowledgable expert in IVR solutions, payment processing and proper PCI-compliance systems-engineering to aid us by building a complete solution using Asterisk and other open-source projects to create a full secure (and PCI-compliant) IVR payment system similar to the commercially available solutions.
Skills: Asterisk Authorize.Net Development Elastix FreePBX
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
We have already configured a Asterisk 13.6.0 with FreePBX 12.0.76.2 in a Ubuntu 14.04.3 LTS (cloud server in Linode) but we need to finish setting some things like the following: -The biggest current problem we want to solve is that external calls are working properly, now sometimes when we make a call to the external number the other person listens to us but we can not do it and sometimes it happens the other way around, so we researched and apparently the best solution is to configure the NAT of our asterisk or maybe some other solucion would be to configure a server STUN or ICE -Fax Configuration (spandsp) -Queue configuration as completely as possible. ... We have already configured a Asterisk 13.6.0 with FreePBX 12.0.76.2 in a Ubuntu 14.04.3 LTS (cloud server in Linode) but we need to finish setting some things like the following: -The biggest current problem we want to solve is that external calls are working properly, now sometimes when we make a call to the external number the other person listens to us but we can not do it and sometimes it happens the other way around, so we researched and apparently the best solution is to configure the NAT of our asterisk or maybe some other solucion would be to configure a server STUN or ICE -Fax Configuration (spandsp) -Queue configuration as completely as possible.
Skills: Asterisk FreePBX SIP Ubuntu
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
I'm looking for someone that knows the Kolmisoft (x8) switch to set a couple of clients (4), a supplier (a2z voip destinations, premium and standard rates) and some DIDs This needs to be done as soon as possible. I will provide the information on the suppliers and clients and on the DIDs' settings
Skills: Asterisk VOIP Administration VOIP Software
Fixed-Price - Expert ($$$) - Est. Budget: $150 - Posted
I need a well skilled freelancer to support WEB RTC on of the Asterisk i will provide. Deliverable will be updated Asterisk and a web page to test.
Skills: Asterisk WebRTC
Fixed Price Budget - Intermediate ($$) - $10 to $350 - Posted
We need to setup and e-mail2fax solution for our cloud based pbx. Fax should be work on T38 protocol aka FOIP. The idea is pretty simple and there are a few old opensource scripts out there. server will accept an email with format faxnumber@ourfaxserver.com and then sent the attached file (mainly pdf or jpeg, ability to do also office files is a welcome) We can provide a test setup VPS server.
Skills: Asterisk A2Billing FreePBX Linux System Administration
Fixed Price Budget - Intermediate ($$) - $200 to $500 - Posted
A base ubuntu box will be provided I need to be able to do the following 1) Setup 2 sip trunks to my suppliers - this will be used as the agteways for the clients 2) Configure clients - the clients will be setup as trunks on my asterisk pbx's 3) Showen how to setup routeing for cost savings 4) Showen how to pull cdrs for billing purposes 5) The gateway will need to hangle the G729 codec, instructions on setup and liceinsing must also be provided 6) What changes have to be made for when i move the machine to site with new ip addresses etc 7) I would like to know how to see active calls through the system. ... A base ubuntu box will be provided I need to be able to do the following 1) Setup 2 sip trunks to my suppliers - this will be used as the agteways for the clients 2) Configure clients - the clients will be setup as trunks on my asterisk pbx's 3) Showen how to setup routeing for cost savings 4) Showen how to pull cdrs for billing purposes 5) The gateway will need to hangle the G729 codec, instructions on setup and liceinsing must also be provided 6) What changes have to be made for when i move the machine to site with new ip addresses etc 7) I would like to know how to see active calls through the system. Why im doing this I currently have 50+ asterisk PBXs And i have 2-3 trunks setup on each oneand im doing the routing at the pabx level. this is becoming way to hard to manage. ... What will need to be tested before payment 1) Setup a trunk on a test pbx system. 2) From a call from my asterisk system i must route the call through the different providers 3) I will need to see the cdr;'s for testing 4) While doing the calls i would like to see live view of active calls If you have any questions please ask
Skills: Asterisk SIP VOIP Administration VOIP Software
Hourly - Entry Level ($) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
I am currently having two separate problems. 1) When i make calls from external to the grandstream there is no audio 2) All external SIp calls are terminated after 32 seconds I am currently using NAT and NOT using the standard asterisk port of 5060 or standard RTP ports (currently using 10,000-11,000) I need some help configuring the firewall to enable the SIP properly on the firewall.
Skills: Asterisk Firewall Iptables