Asterisk PBX Jobs and Contests

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
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.
Asterisk PBX is a topic that needs skill and if you are an expert in this, then you should earn money through what you know best. There are thousands of jobs posted on related to Asterisk PBX and if you at a pro in this particular field, then will offer you a chance to work on projects you understand. The site attracts some of the best-paying clients and offers an easy-to-use platform, where freelancers can browse and bid on jobs they are interested in. You can simply start your career in Asterisk PBX at today.
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Project/Contest Description Bids/Entries Skills Started Ends Price (USD)
Asterisk client using WebRTC - 2 We currently have an application under WebRTC architecture. We need to transform the audio/video calls compatible with Asterisk IPBX with recording included for: Audio and Video calls and recording. 14 Java, Javascript, Asterisk PBX Aug 17, 2017 Today6d 23h $1130
ubuntu and puppy LiveCDCustomize os i need a os Ubuntu and puppy u can make live CD you have to install some pakage openvpn asterisk php teamview 9 fastd ntp iptables iso size must be under 450MB if you can make it please bid 6 Linux, Software Architecture, Asterisk PBX, VoIP, Ubuntu Aug 17, 2017 Today6d 21h $226
Connect FreePBX CID Superfecta to MS SQL We would like to do a lookup on incoming calls so the client name shows. Lookup would find the number in the MS SQL database and FreePBX would show the client name along side the number. I will provide the layout of table and you should provide instructions on how to set everything up. 4 Asterisk PBX, Coding Aug 17, 2017 Today6d 19h $169
VOIP and AsteriskPBX developer Small task which needs with the configuration of VOIP sessions. Everything is being done at the server level. Need someone who can implement handler at the front end. Deadline =7-10 strict days. 10 PHP, Asterisk PBX, VoIP, HTML5, CSS3 Aug 16, 2017 Today6d 12h $230
VOIP developer required. *** URGENT*** I need someone who is an expert in terms of implementing VOIP services in websites. Its a small task with a strict deadline. Please bid if you seriously know how to implement VOIP. Deadline is 10 days max. All the backend has been implemented already. See you in chat. 7 PHP, Asterisk PBX, VoIP, HTML5, CSS3 Aug 16, 2017 Today6d 12h $152
SIP based VOIP Calling application A SIP based VOIP calling application that will require both (client) android and Server (VOIP Server) as well as Web RTC 40 Mobile App Development, Asterisk PBX, VoIP, FreeSwitch Aug 16, 2017 Today6d 6h $4413
ictdialer installation need to install ictdialer on centos 5 Asterisk PBX, MySQL, FreeSwitch Aug 16, 2017 Today6d $20
Complete voip solution Telecom We do what we want ? We we want to the best of our knowledge. We don't know about us, please tell you. Direct us. We want a complete Telecom solution. So in asterisk that can do everything we want. For example, centos + asterisk + php + nginx + mysql + a2billing + freepbx multi-tenant and other service components to the server must be established. To run mysql on a separate server for the web site hosting services on the server will work on a backup basis. Asterisk on a separate server and other services will work on a backup basis. we're thinking of using VMware ESX virtualization software. At the same time security is very important for us. We want to know suggestions and what you can do about security. For example, VPN with SMS authentication entered entries or double. The software can only be used by us that will allow you to security measures need to be taken. So, another person unauthorized use of the software. We need to be able to communicate to the server as the script and ISO software. Of our work is 100% solid, we want to make. For example, the data base will be located on two different servers and the same information is supposed to be there. So made redundant. On the asterisk side in the same way, we want a redundant structure. In terms of VoIP services, there should not be an interruption in service. We work with a global Tier 4 certified data centers. So continuity is very important to us. The following is my personal opinion on redundancy and security. E5 cpu + ram RAID 10 hard drive 4 nvme ssd 64 GB 7 Centos + cpanel + dns and MySQL cluster + cdn + cloudflare security. The SRV record. The closest location geographically (MS ping) the Asterisk server provide quality service to the customer. Other routines use Trunk routes, in case of failure. And we want to use the cheapest route. iphone + android + mac + Windows voip we can think later of their software. But we can evaluate the proposals. The colors of the site design website design VoIP with the drawing visually the same, and it must be very beautiful. For example, blue and light gray colors and other compatible primary colors, sub-colors. For example, graphical Dashboard screens of the graphic display of the report and the CDR is supposed to be good a lot. Changing the menu position that we see as unnecessary of removing unused features. We want a responsive design. Whmcs automation software do I need to use ? if you need whmcs to be integrated into the system and eat all we want. Did we want the integration to be performed from the service providers api. Customers before they reach us easily via PayPal, Credit Card payment services such as we want to load balance automatically using. We have to work with the person or company long-term, we want to do accurate and honest work. We leave it to you, do not leave us. We want you really to rely on. We know the prices on the market. We are awaiting a quote logically correct. Us make suggestions. This way would be better. It is the truth. If we do what happens in the future. Would be better or worse. Us just need to be honest. Long-term support and maintenance agreement we can make. Written training documents as we may require a video visual for our customers. Long-lasting detailed communication will make the decision after open discussion. What can you do for us ? How long will This take ? What is the price ? How to pay with ease and what can we do ? If you have questions, please ask. Hopefully what I want to tell right description. 12 PHP, WordPress, Asterisk PBX, VoIP, HTML5 Aug 16, 2017 Aug 16, 20175d 23h $16
Web Gui for Chan Dongle Gsm Gateway The goal is to establish an asterisk server for the purpose of voip termination in GSM network. The freelancer has to do the following: - Install asterisk server and configure it to operate with Huawei usb dongles (modems) having prepaid simcards to connect to the GSM network. - Be able to attach more huawei usb modems to the server if needed. USB hubs will be used to attach the usb modems. - The server shall be able to terminate voip calls comming in codecs g729 and g723. - Send ussd and ivr to each sim card to topup and credit check. - Display how much credit is available in each sim card. - be able to listen to the channels if they are busy. - If a channel is busy the call shall try automatically the next channel and so on till it finds a free channel. - GUI interface to monitor and configure the system -sim card suspension detection based consecutive unsuccessful number of calls and highlighted with red color -sim card suspension detection based on sim card out of credit - automatic topup sim card when reach value like 1$ - show dongle sim number - show hub number the usb dongle is connected to, to keep track of it. - ability to send ussd code on a selected number ( click dongle and have options to send ussd and sms, and read sms inbox) - ability to remove dongle in gui or stop it ( remove now or stop now ) - restart dongle and send code again. - Human Behavior Simulation : random sms, and calls between dongles. - set dongle call duration limit and time swap. - CDR stats by dongle: duration, Acd, ASR... - sms read ( ability to decode the non readable sms in CLI) - dashboard of live stats (active calls, acd, asr, calling dongles.. sell number...) - show antenna mast (BTS) ID the dongle is connected to. - random rotation of calls to minimise blocking sim cards. - database to store dongle imei and imsi of simcards an relationship between them. -auto remove non present dongles. - GUI to allow enter prepaid sim cards codes to database for automatic recharge later - hide/show numbers - Module to change IMei number of dongles You are free to specify which distribution and version of Asterisk to be used. You shall define which model of Huawei usb modems have to be used with asterisk (compatible with Linux and asterisk) 20 PHP, C Programming, Linux, Asterisk PBX, VoIP Aug 16, 2017 Aug 16, 20175d 20h $579
Configure "Inbound" and "Trunk" on freePBX, settings up from a Twilio SIP Trunk I have freepbx on local machine connected to SIP at Twillio. Outgoing calls are working, Extensions with IP phone and soft phone work, but the incoming needs to be tweaked. I get error: NOTICE[14462]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from (callid: 019c56e2136b92a6bb57603e2cb85afd@ - No matching endpoint found Also extensions are not showing connected or the IPs when showing peers in asterisk. 10 Linux, Asterisk PBX, VoIP, UNIX, Network Administration Aug 15, 2017 Aug 15, 20175d 9h $31
Remove member ID asking for the first time on A2 Billing on Elastix Need someone to make some change to A2Billing on an Elastix server to make possible a caller to call a member without having his member ID so make the system recognized the number that is called ( a member one) and then make the call through. As system already recognized a number the second times it calls, need to make it the first time and add this number to the member's list of DID autorized for him. 10 Asterisk PBX, MySQL Aug 14, 2017 Aug 14, 20174d 3h $171
Asterisk Freepbx dynamic text to speech ivr When calls come in, we want the ivr to say "The number you have dialed #### is on the line, please hold while we try to connect you.". The #### is the number dialed and we need Freepbx to dynamically convert text to speech for the user to hear. You need to configure as well as teach us how to configure. 13 Windows Desktop, Linux, WordPress, Asterisk PBX, Windows Server Aug 14, 2017 Aug 14, 20173d 19h $122
Fix 'Allocation Timeout' Error in TURN Server Coturn is a TURN server and turnutils_uclient is a test client for Coturn. Testing Coturn with turnutils_uclient on a CentOS or Ubuntu server results in 'Allocation Timeout' error in the Coturn log. Attached are logs from tests carried out using the test client and Coturn, for UDP and TCP connections. POSSIBLE CAUSE: Checking netstat command shows that there is a duplicate UDP interface/connection for each tuple combination. For example: [root@server1 log]# netstat -lnptu | grep turnserver tcp 0 0* LISTEN 31552/turnserver tcp 0 0* LISTEN 31552/turnserver udp 0 0* 31552/turnserver udp 0 0* 31552/turnserver Rebooting the server or killing the process does not remove the duplicate interface/connection. It may be possible that this causes the allocation timeout, however this is not confirmed. Coturn server: [url removed, login to view] Test client: [url removed, login to view] 10 System Admin, Linux, Asterisk PBX, VoIP, Network Administration Aug 14, 2017 Aug 14, 20173d 15h $129
Fix issue with Vtiger Freepbx Connector After installing and configuring Freepbx and Vtiger Connector, I am not able to get it to work. 3 Asterisk PBX, vTiger Aug 13, 2017 Aug 13, 20173d 6h $92
Asterisk Expert We are looking for long-term asterisk experts on hourly basis. We need very strong developers with excellent hands-on experience and debugging abilities. A senior developer with design skills and aspiring to be an architect. You will have to sign an NDA. 14 PHP, Asterisk PBX, VoIP, MySQL, FreeSwitch Aug 13, 2017 Aug 13, 20173d 2h $11
Asterisk / TrixBox support Need to make changes to extensions and multi-ring settings on various extensions for very old system with polycomm phones 13 Asterisk PBX Aug 11, 2017 Aug 11, 20171d $21
Administrador VOIP/Linux/Asterisk Administrador de Telefonía VOIP experiencia en Asterisk comprobable, Linux , programación LAMP. 31 PHP, Linux, Asterisk PBX, Apache, MySQL Aug 10, 2017 Aug 10, 201712h 34m $659
Ringless voicemail drop software development with billing capacity Leave voicemail without ringing the phone. You must have previous experience on rignless voicemail drop Please present your previous work you have done only get paid after delivering the job 11 PHP, Asterisk PBX, VoIP, FreeSwitch Aug 10, 2017 Aug 10, 201711h 58m $196
Asterisk CDR Tweaking would need Freepbx CDR Hacking.. we have external CRM and need to fetch via cron which calles we have missed and which calls we made... cronjon in php works, but there are so many duplicates entries from queues in CDR, etc so its very hard for us to find out which call was missed, etc. hope you get my point. thanks 8 Asterisk PBX Aug 10, 2017 Aug 10, 2017Ending $61
Simultaneous Caller Phone I need a program which can dial numbers simultaneously and over and over and over again. List of voip providers would also help in this scenario 21 Software Architecture, Asterisk PBX, VoIP, Telephone Handling Aug 9, 2017 Aug 9, 2017Ended $1250
Write some Software I would like to build some software that will call a phone at least 3 times a minute. I would like it to be compatible with my Skype. That means I would like to connect my Skype to the software so I can use the credit to make calls. Please send a message beginning with the code: 7844 if you would like to help with the idea. I know this is a very small task as ive been offered a software bigger then this for $11. 7 PHP, Software Architecture, Asterisk PBX, Telecommunications Engineering Aug 9, 2017 Aug 9, 2017Ended $22
configure vicidialer I need someone to quickly configure some of aspects of my vicidialer. Below are exactly what i need setup. * Setup 2 inbound DIDs * setup 3 campaigns, making giving them the ability to do inbound and outbound, giving inbound priority on one of the campaigns * Setting up 3rd campaign to be a survey campaign (voice broadcast) * setting up ivr to route to certain extensions or specific campaign * setup voicemail * setup carriers for inbound and for outbound * making sure the firewall is good so that i dont get hacked. 10 Asterisk PBX, Call Center Aug 9, 2017 Aug 9, 2017Ended $154
Fix no voice heard currently only person in conference in vicidial Fix no voice heard currently only person in conference in vicidial 7 Asterisk PBX Aug 9, 2017 Aug 9, 2017Ended $117
Simultaneous Calls/Phone Flood Hello Candidates. I need a program which i can make simultaneous calls as well as redial numbers multiple times. Now i know there is some providers which allow and some which do not. I need help with making sure this program works and that the provider will be agreed with such a system of calling. thank you 15 Asterisk PBX, VoIP, Software Development, Telephone Handling, FreeSwitch Aug 8, 2017 Aug 8, 2017Ended $1608
Robo Dial/ Phone Flood Hello Candidates. I need a program which i can make simultaneous calls as well as redial numbers multiple times. Now i know there is some providers which allow and some which do not. I need help with making sure this program works and that the provider will be agreed with such a system of calling. thank you 4 Software Architecture, Asterisk PBX, Telecommunications Engineering, Telephone Handling, FreeSwitch Aug 8, 2017 Aug 8, 2017Ended $384
Get the incoming Caller-ID from Asterisk PBX and show it in a pop-up screen once the call coming We have .NET Restaurant Management System. this system has a module which handles clients' orders through the phone. we Have Yeastar PBX (different models) which is Asresik platform. we need to connect our system with PBX. So once incoming call happens a pop-up screen appears with caller-ID 14 Asterisk PBX, VB.NET Aug 8, 2017 Aug 8, 2017Ended $198
Need Asterisk and freepbx Support hi, we need asterisk support. only experienced person can apply. thanks. 17 Asterisk PBX Aug 8, 2017 Aug 8, 2017Ended $148
VICIDIAL Asterisk Pass CLID to Agent Softphone VICIDIAL Pass CLID to Agent Softphone - Dial Method Auto/Predictive Dialing - Agent is in a conference technically. therefore vicidial doesnt pass the CLID to the softphone. - We can show on teamviewer what we need. It should be a simple project. we need to find a way to pass every call that is getting connected with agent to pass CLID to Softphone. 10 PHP, Linux, Software Architecture, Asterisk PBX Aug 8, 2017 Aug 8, 2017Ended $133
VoipSwitch Setup Hello, Do you have experience voipswitch latest version 3.0+? If yes we need your help to setup our new switch. Kindly post your bid Thank you 7 Linux, Asterisk PBX, Software Testing, VoIP, FreeSwitch Aug 8, 2017 Aug 8, 2017Ended $13
Project for APCTelco Hi APCTelco, I noticed your profile and would like to offer you my project. We can discuss any details over chat. 1 Asterisk PBX, VoIP, , FreeSwitch Aug 8, 2017 Aug 8, 201717h 24m $13
install and configure freepbx sangoma system and fix issues ONLY PBX EXPERTS PLZ install and configure freepbx sangoma system and fix issues with existing installations ONLY PBX EXPERTS PLZ 12 Asterisk PBX Aug 7, 2017 Aug 7, 2017Ended $16
Cisco phones directory with asterisk based pbx trying to connect cisco 8811,8861 and 8945 with directory and asterisk. 8 Asterisk PBX, Cisco, Active Directory Aug 6, 2017 Aug 6, 2017Ended $135
elastix elastix pbx 14 Asterisk PBX, Cisco, Debian, FreeSwitch Aug 5, 2017 Aug 5, 2017Ended $369
LDAP manual for Gigaset base I have a Gigaset N510 IP Pro DECT. Under Settings -> Directory -> Corparate Directory, there are settings to use LDAP to lookup a caller's name. I am looking for a manual how to config my N510 and how to config LDAP (using) OpenLDAP, so I can see the name of the caller. The manual must be a step-by-step how to config the N510 (eg. a screenshot of the settings) and how to install and configure the LDAP so my N510 can communicate with it. I also would like a short discription how to edit add entries within the LDAP and add new callerobjects. I would consider the project finished, if I can reproduce a working system with the manual. I am a IT professional so the explanation e.g. Go to the folder D:Test is sufficient for me. It doesn't need it to be: Go to start, type Explorer, double clcik on Explorer, click on Computer, double click on drive D:, double click on the folder Test. The LDAP settings of the N510 seems to be the same for other Gigaset base stations like the Gigaset N300A IP DECT, the Gigaset DX800A, Gigaset GO-Box 100. If your Gigaset base can be remotely administered with a webbrowser, it probably has this LDAP function to lookup a caller's name. Later on I will provide some more information (links) and screenshot(s). 4 Asterisk PBX, VoIP, Anything Goes, Telecommunications Engineering Aug 4, 2017 Aug 4, 2017Ended $270
Conference Call With Asterisk version 13 or Asterisk version 14 I will like to use either of the asterisk versions specified in my title above to get a conference chat room working with the following features; a. The Chat room must have two (local) extensions. One for the "Internal Participant" and another for the Admin. b. Persons (from PSTN) who want to join the conference will call the Admin extension and he will add the caller to the chat room. Only Admin can add people to chat room. c. Is there a way to monitor all channels in the chat room in real time to either remove or mute any channel for whatever reason including a noisy channel. d. Unlimited Participants expected Regards 17 PHP, Asterisk PBX, VoIP Aug 4, 2017 Aug 4, 2017Ended $155
Vicidial Auto block number if called 4th time in 15 minutes Hi, i have dailer of vicidail 2 server based on cluster i get to many prack calls i want to block them auto in dailer if any caller called as 3 time in 15 min it will be auto block by dailer 14 Linux, Asterisk PBX, VoIP, Call Center Aug 3, 2017 Aug 3, 2017Ended $325
I would like to hire a Cisco Engineer . 28 Asterisk PBX, Cisco, VMware, FreeSwitch Aug 2, 2017 Aug 2, 2017Ended $130
Predictive dialer for freepbx - asterisk I need a predictive dialer that can diale several numbers at once, and put them to a freepbx queues DIALER The dialer will call X number of customers, depending of how many free agents and the droprate. When the customer answer, the dialer will put the call to the queue. The dialer will detect busy, no answer and answering machines. GUI I need a simple gui for controlling some parameters: - max lines to use ( for all agents) - max lines to use for one agent - droprate % - Answering Machine Greeting Time MONITORING/REPORTS droprate in % avarage waiting time number of no answer and rate 10 Asterisk PBX, CRM, Desktop Support, Business Analysis, MySQL Aug 2, 2017 Aug 2, 2017Ended $675
Write some Software A project to build a add-on for call center such as Screen-pop, FreedomQ, FCR, Analytic, etc. 9 Asterisk PBX Aug 2, 2017 Aug 2, 2017Ended $14
Virtual pbx (backend and frontend) -- 2 The project is to build a full virtual PBX system, both the back-end and front-end. instead of writing all required features I would like you to see evoice . com This project will be to create the same system. Basically I would like clients to register and pay Once they register they can transfer their existing number from us or buy a new number from us. After that the client can configure extensions as they require by accessing their account on the website. They can also specify where these extensions should ring. Plus every feature provided by evoice . com You would also need to advise what hardware and other equipment we would require. 1 Asterisk PBX, FreeSwitch Aug 2, 2017 Aug 2, 2017Ended $4433
OpenSip/Asterisk DB Configuration via OpenVpn(30-400) Please READ Carefully before you bid. We have 2 School campus at different locations We have Asterisk A2b as central DB billing it has static public IP We have 2 OpenSip Servers, one at each school location. These Servers does not have Static Public Ip's. They are behind Firewall. Each Server are given local Ip from a wifi network at each of the school. (these Server can see outside but you cannot see them from the Net) To Link all 3 servers we used OpenVpn sevices. So now each of the Server has 2 set of Ip's 1. A2b has a Public IP to communicate with anyone and openVpn IP to communicate with any of the OpenSip Servers. 2. Each OpenSip Server has 2 ip's. One local which sip clients use to register to it from the Wifi network. And 1 ip from OpenVpn which it use to comminicate to A2b or to route call to the other OpenSip Server. What we want to achieve. Students can call anyone registered to any of the servers Free (at no additional internet cost to them if they register from the wifi network) we the Company will charge small daily or weekly or periodic fees But calls are FREE to Students when they call anyone register to any of the Servers When they make Out of Network calls then we bill as normal via rate tables 13 Asterisk PBX, VoIP, MySQL Jul 30, 2017 Jul 30, 2017Ended $149
Elastix on hyper-V Need to be add hardware on Hyper V 4 Asterisk PBX Jul 29, 2017 Jul 29, 2017Ended $32
Connect 2 IP phones in different countries I would like to connect 2 IP phones, one in Saudi Arabia and another one in Egypt. Simply, I need to make to calls between these 2 IP phones to avoid the high charges on international call. I need to know if possible? How can I know if my internet provider is not blocking such feature? If possible what is requirements: server, internet connection.....etc? I will need a technical guidance on how to install the system. 12 Asterisk PBX, Phone Support, Network Administration Jul 29, 2017 Jul 29, 2017Ended $153
Asterisk Freepbx dynamic text to speech ivr When calls come in, we want the ivr to say "The number you have dialed #### is on the line, please hold while we try to connect you.". The #### is the number dialed and we need Freepbx to dynamically convert text to speech for the user to hear. You need to configure as well as teach us how to configure. 16 Asterisk PBX Jul 29, 2017 Jul 29, 2017Ended $126
Project for dcapape Hi dcapape, I noticed your profile and would like to offer you my project. We can discuss any details over chat. 1 PHP, Visual Basic, Asterisk PBX, HTML, jQuery / Prototype, Jul 28, 2017 Jul 28, 2017Ended $12
Project for Asterisker Hi Asterisker, I noticed your profile and would like to offer you my project. We can discuss any details over chat. 5 PHP, System Admin, Asterisk PBX, VoIP, , VPS Jul 28, 2017 Jul 28, 2017Ended $38
Basic FreePBX / Asterisk configuration Hello, I'm looking to implement FreePBX for my business. It's a smaller operation (2 locations) with only a few employees. Only 5 or 6 extensions are needed. I have installed and partially configured FreePBX on a web server and created a trunk for The system is partially working, but I cannot get my softphones to connect. I'm certain it's a port / firewall / permissions issue, but haven't been able to figure it out. I tested out Zulu since it comes prepackaged with FreePBX and has a free 2 seat license. (again only using this as a test) but the Zulu clients connect just fine. There is only 1 incoming phone number (DID), and I can receive calls just fine in FreePBX, but dialing out from Zulu does not work (it just rings and goes to FreePBX voicemail). Here's a list of the main things I would like you to do: - Create 5 extensions (i.e., 100/200/300/400/500) - Incoming calls should be answered automatically by auto attendant with music on hold (I'll create a recording that says "please wait for assistance, someone will be right with you", then shortly thereafter "please continue to hold, or press 2 to leave a voicemail"). If nobody answers within 60 seconds, the call will be forwarded to my cell phone - figure out the correct softphone configuration and confirm that calls can be sent and received by softphone (using whichever softphone supports the highest quality codecs) - Ensure that FreePBX is configured to use the highest quality codecs available (limited by our [url removed, login to view] provider, of course) - Harden FreePBX security - Is it recommended to set up a VPN to connect to a remote FreePBX installation? If so, we'll need to configure OpenVPN as well (I know how to configure the OpenVPN client on computers, but not sure how the physical phones would know to use the VPN) That's basically it for the moment. Let me know if you have any questions. Only high quality providers with lots of 5 star reviews will be considered!! If you have negative feedback and/or low feedback, please don't bother bidding. Thanks! 26 Asterisk PBX Jul 28, 2017 Jul 28, 2017Ended $190
Vicidial and Goautodial keeps crashing We have a system with 100 agents and it keeps crashing. We configure our system with the following versions in order to use the web phone. Now we are experiencing many crashing issues such as: Database corruption Asterisk Completely dropping Sockets going from required 9 to 6 Time synchronization Vicidial Version, VERSION: 2.14-617a Asterisk Version,11.22 CentOS Version, CentOS Linux release [url removed, login to view] GoAutoDial Version, 3.3 AstGuiClient Version ,2.14 13 Asterisk PBX Jul 28, 2017 Jul 28, 2017Ended $56
Asterisk PBX job 1- Configure PBX eth3 with a Separate Network to PBX eth2 pre configured External Network so that both networks are isolated and users on both networks can make calls in and out fine. Hardware is sorted only configurations on PBX and Linux. 2- Configure ability to pickup incoming calls from any handset even if it isn't ringing. (i.e reception phone is ringing and one of offices hears phone ringing and pickup). 3- Check is any error and fix. also help me change some passwords. 19 Linux, Asterisk PBX, VoIP Jul 28, 2017 Jul 28, 2017Ended $16
Asterisk call center Setup small customer support call center with 8 extensions, isdn and gsm out, hardware is ready. 11 Asterisk PBX, Call Center Jul 27, 2017 Jul 27, 2017Ended $558
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