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    5,000 asterisk pbx jobs found, pricing in USD

    Seeking to have an API developed to be able to read and write presence data into Office 365 Skype for business. Eg when user is on call like to set external pbx busy lamp to busy & reverse presence also required as in if on external pbx call require to set Lync/Skype for business presence as busy. Project needs to support Office 365 version of Skype for business as well as on premises version.

    $1438 (Avg Bid)
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    6 bids

    i have install freepbx on rasqberry . i have a sip number from a provider and i want install my sip settings on freepbx to do and receive calls on my 3 local extensions . my local extensions is ready , i need only sip configuration for inbound and outbound route .

    $26 (Avg Bid)
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    I have one server of Elastix and i want to enter 20 IP that i have one list, and 4 outbound IP. All ports have close and the server have to secure.

    $31 (Avg Bid)
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    Currently have a working FreePBX install on Sangoma hardware. The hardware is old and not rack mounted. I have new rackmount hardware (online, ready, with IP address, free PBX installed etc) the following needs to happen to it; Port the current working config to the new hardware. Re-configure the phones to point to the new hardware. The phones are Yealink T28P phones On the handsets, make the BLF fields light up as if their line keys (Line1 Line2 etc - 1 per SIP trunk) - i have 5 phones, if you can make one work i'm sure I can copy the config. Configure the GSM card (Sangoma W400) - The new hardware has a 2 port GSM module in it, which will be used for voice and sms, so configure the SMTP reciever and the voice sims as a trunk etc. Configure the 2 analogue trunks & 2 anal...

    $147 (Avg Bid)
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    I would like to order a setup of Zabbix server which will be dedicated for automatic PBX monitoring within a separate VLAN. Measured values and triggered actions below: 1) Disks, PBX operation threats for HDD overload. 2) ETH analysis, if there is a bigger load of interface than the declared size – start a fraud alarm; 3) RAM analysis, PBX operation threats for RAM overusage. 4) Simultaneous connection analysis, threats of PBX operation under the scope of overload of simultaneous connections (filling the purchased bandwidth in Zabbxi, alarm at 75%); 5) Is the phone available, meaning whether the device is available wthin network, continous statistics of device operation, whether the DND option is enabled, forwarding, if DND enabled – is it possible ...

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    We have Asterisk server which can handle WebSocket protocols. Our task is to connect JSSIP to the ready design and layout for the work of the site. Documentation reference Http:// Links to the library: Https:// Demo: Objectives of the project: -User login with defined credentials. -Providing a point-to-point call through the browser. -Providing chat. -Presence function. -WebRTC Video Quality configuration (with predefined qualities and video camera and microphone switching) . - Indication of ringing and busy state. -The integration code must be copyright, not stolen. Layout: -crossbrowser, cross platform adaptive layout -adaptation for mobile devices (recommended bootstrap layout

    $273 (Avg Bid)
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    Create a Program that would help a student pick the classes throughout college until he/sh graduates prerequisites must me taken into consideration i.e if cosk1020 is a prerec for math2070 math2070 cant be completed until cosk1020 is on check sheet one asterisk is prerequisites 2 asterisks is co-requisites

    $39 (Avg Bid)
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    Create Application or Web based app get list of users from database call users on order log call status (success,failed,answered, action taken) press key send http request set number of simultaneous calls

    $157 (Avg Bid)
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    Minor changes including: - Adding a recording if the user doesnt dial the right number on their phone. - Letting the user try again. -If no # dialed on the phone and hung up return a 0 to the php based dasboard.

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    please find below project requirement which we need to add in the fusion pbx 4.0 running on free switch.1. Project Scope:Enable API setting to send inbound caller ID calling party information for any domain/tenant as fully supported solution.2. Use case:Client want to send a caller id information through API to Icabbi for domain configured inside fusion pbx tenant domain name olympiacentrewest.local3. Example scenario:Calling party with CLI 075072378510 calls inbound route for DID 02073867810 Call will ring any extension/ring group/queue/external DID pbx will also send calling party number through API as shown in below link with international Enum format  447507237810 and call extension number example URL is as follow

    $171 (Avg Bid)
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    I have "Asterisk 13.1.0~dfsg-1.1ubuntu4" its been working fine as my private voip server. However I would like to use the sms / messaging feature. I do not have any GUI or 3rd party addons this is a stock Ubuntu Asterisk install. I am using Zoiper on Android and would like to use the SMS / messaging feature. Also my DID supports incoming/outgoing SMS so would like that to work. I would like if possible some one to guild me through what I need to do to make this happen. I cant give access to the server but can give the files u request (less any private info in them) Please dont just point me to some readme u googled as I have tried a few with out luck. So this will most likely need some one with experience.

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    I have a voice blast software with code written in asterisk and Php. It is integrated with an application and on hangup of the call, i am sending a hangup message to the application. The code functionality is working fine except in one scenario. While the call is been executing, if the customer disconnects and it moves to GoSub return context , in this particular scenario rest of the dial plans are not executing. It will hang up the local channel log for SIP/Dahdi channel hangup so i am not able to trigger the hangup message to the application. It is an urgent requirement to be started today. Only bid if you are an expert on asterisk dial plan.

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    please find below project requirement which we need to add in the fusion pbx 4.0 running on free switch.1. Project Scope:Enable API setting to send inbound caller ID calling party information for any domain/tenant as fully supported solution.2. Use case:Client want to send a caller id information through API to Icabbi for domain configured inside fusion pbx tenant domain name olympiacentrewest.local3. Example scenario:Calling party with CLI 075072378510 calls inbound route for DID 02073867810 Call will ring any extension/ring group/queue/external DID pbx will also send calling party number through API as shown in below link with international Enum format  447507237810 and call extension number example URL is as follow

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    10 bids

    Hi, We have a Live Asterisk panel that need bug fixing.

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    15 bids

    ive got freepbx on a server and am using paid version of vtiger. need to configure the asterisk credentials in vtiger. 20 minutes of work.

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    We have an extension which has a voicemail box with VMX locator on which option 1 get's forwarded to a cell phone. We need when the caller presses 1, the call should go to cell phone and the recording of the call should get emailed to a group of users.

    $53 (Avg Bid)
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    I'm looking to develop a router capable of supporting IPSEC/OpenVPN/MultiCast to serve as a VPN between a local PBX and a remote site. The VPN and PBX will be hosted in a Xen Hypervisor network. Goal is to allow the VPN server to maintain remote VPN connections so phones and computers can communicate back to the hosted pbx without ports changing, etc.... I would like to use existing open source technology to keep deployment costs down. At the end of the day someone with great experience with VyOS, pfSense or another comparable router and VPN would be perfect.

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    From an Apache host create a simple PHP website that connects to a separate Asterisk software PBX and shows a real time display of the extensions that are in use and displays incoming called ID. It would need to display the Extension, user, status, incoming/outgoing, number and directory entry in a simple table format. It would also show the incoming number / directory entry for any incoming call separately. The idea being that if required this could be displayed on a large monitor so that users can at a glance see the status of each user as well as caller ID for incoming calls. There is a LDAP directory on the Asterisk system numbers need to be compared to this so that the directory entry is shown instead (or maybe as well as) the number. A demo system wo...

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    We need someone to show us how to provision phones with Freeswitch and Asterisk if possible. We have a running in production FreeSWITCH server, just need to work on getting it setup. Same goes for FreePBX servers as well. Phones are mostly Cisco but we can work with any brand based on recommendation. Additionally if you have the skillset we need you to integrate kamalio in our current production system or into a new environment where we can move our customers going forward.

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    Working Chan_dongle box I can see incoming sms and ussd in cli but they do not come in log file. I think it is some mistake in

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    hey, I have an old version of an opensips server not use, its just there incase i need it, its infront 2 asterisk server was using for LB and sip registration, wanted to update it to the new ver of opensip can you do it?

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    Need to be able to live transfer call with originating caller id. Part 1. Agent will call Customer, and tell customer we can help with service and will transfer to Specialist. Part 2. Agent dials specialist while customer still on line. Specialist answers, we make introductions between customer and specialist, and disconnect from call. Allowing Customer and Specialist to continue conversation. Part 3. Very Important: When Agent is dialing specialist, Caller ID must show Customer Caller ID. Part 4. Very Important: Must be able to have access to entire call recording, from initial call from Agent to Customer, Agent to Specialist, then Specialist and Customer full conversation even after Agent hangs up. Thanks! Happy Bidding Rob

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    We need to configure one server which can accept voip call and terminate on a CENTOS with Asterisk 11 sip tls . authentication is done on cli basis

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    We have Asterisk 13 using FreePBX 13 and VTiger 6.5 source I can make outbound calls from vTiger but I can't get the inbound calls popup showing on vTiger for users. Also I need the PBXmanager populated with the CDR details only for user of vTiger not populate for all calls into Asterisk Have TEST system ready with Asterisk, vTiger and FreePBX for you to make the changes. Require all source code and instructions on what changes were made so I can make the same changes to our live system

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    hello guys. who can install us asterisk, german admin interface and visal pbx manager that a beginner can handle everything? its important that asterisk is secure and not be hacked in 2 minutes :)

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    develop apps with asterisk pbx

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    ...configure it so the Asterisk reload over night. Also nee to install Voice Recognition (CMU Sphinx) for IVR. Dialplan for IVR will provide the option for Customer based on some criteria than send the caller to custom destination which will work with the voice recognition to ring the requested group. Allow user till 3 attempt, if it fails than it will ask user (announce) to enter 3digit service code or press defined character to rout the call to an agent. - I will set the service code for the user and display those in a directory. Check the attached document before you submit proposal. Please be specific or your bid will be rejected. There will be more task if you can accomplish this primary task. **** Submit proposal if you have hands on experience in FreePBX, Asteris...

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    Phone has 8.5.4 SIP firmware already. Need XML to enable features for BLF, parking, speed dial and NAT. Need TFTP provisioning instructions. SIP Server is a commerical asterisk implementation.

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    Hello, I have one audiocode mp118, I would like someone to set up it with my asterisk I have already the sip account, I am looking for someone know how to setup the audio code best regard

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    Hello, I have one audiocode mp118, I would like someone to set up it with my asterisk I have already the sip account, I am looking for someone know how to setup the audio code best regard

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    ...less than 3 they have must be interested in expanding) We need to know who their current provider is. We also need them to provide a copy of their last months bills so we can provide them with a comparison quote. Product: We can typically save our prospective customers between 35-60% on their current call spend. We also offer the following features as part of our package: Hosted PBX Call Recording Call Monitoring Voice-mail to email Soft App SIP Trunk Support (Remote and customer service) Number porting. ...

    $952 (Avg Bid)
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    ...ar/ I need someone to configure my fresh freepbx server, i have no skills to configure how to route internal calls and two lan cards , all trafic goes to one card, and i need to route sip trafic to the eth that connects to my sip provider. Please see graph. My asterisk server has no public ip , so work will be done using my internal pc with teamviewer, anydesk or similar. My needs: 1. configure routes in the server and configure the lan cards to work fine the calls from my pcs to my sip provider, asterisk now is trying to make calls with internal eth 2. configure routes so i can add any internet sip provider to make calls to other countries. So in this case we will add a new outgoing route in freepbx that will be outgoing using internet not telecentro (see g...

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    Hi kingAsterisk, I noticed your profile and would like to check if you can fix some configuration issues we have on our asterisk / freepbx server. currently not able to do outbound calls getting "chan_sip.c: Retransmission timeout reached on transmission" error Inbound and extension to extension calls works Thanks!

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    I'm looking for someone who can create a few TRENDY looking designs for my T-Shirt Company. The ideas I have are as simple as getting a cool way to say: 1) Bride...Bride 2) Team Bride * 3) Bride Tribe * 4)Getting fit for the dress (caption "2017 Bride") 5) Mrs. 6) Squad * 7) Bride's Entourage 8) Warning - bachelorette night in progress 9) Warning - bride You don't have to tackle them all. But state what you would tackle with your bid. PLEASE NOTE: if you are aiming to tackle the "bride" AND corresponding bridesmaid shirt (they are marked by an asterisk) ... please aim to have the design match. I.E using the same font. But different colors are encouraged. The "bride" shirts will be on a white shirt, whereas the correspondin...

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    Hi, we need to implement a provisioning in elastix 4 for cisco 303g2, cisco 502g, cisco 509g, end some grandstream devices for example GWX 4248. In this case, for test, we need to do a lab for 4 ip phone, and elastix, ip phone will be on separate vlan for communicate to this case, for test, we need to do a lab for 4 ip phone, and elastix, ip phone will be on separate vlan for communicate t... we need to implement a provisioning in elastix 4 for cisco 303g2, cisco 502g, cisco 509g, end some grandstream devices for example GWX 4248. In this case, for test, we need to do a lab for 4 ip phone, and elastix, ip phone will be on separate vlan for communicate to this case, for test, we need to do a lab for 4 ip phone, and elastix, ip phone will be on separate vlan for communicate to pbx

    $70 (Avg Bid)
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    Have an old PBX which is a Nortel Option 61c. The unit is showing some errors, on loop 8. I need remote assistance to reboot this card and also assistance to program 3 new analog extensions.

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    We need a module developed for ombutel that would let us integrate it with zendesk so that it provides functions similar to: /apps/five9-for-zendesk/ /apps/asterisk-dtk-by-cdc/ The module should meet the following minimum criteria: - Be installable through the ombutel module admin - Must be installable in zendesk as an app - Must be able to assign zendesk agents to extensions - When an incoming call is coming in a popup should appear in Zendesk containing the following 2 scenarios: 1) if the customer is in zendesk the customer name should appear with an open profile link and reject call link, . 2) If the customer is not in zendesk the phone number should appear with a create contact link. - If the call is answered a new ticket should be made with relevant info already pop...

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    Hi MikeRRR, I noticed your profile and would like to offer you my project. I need assistance setting up Freepbx with Twilio SIP trunk to work properly. The system currently have two problem need to be...profile and would like to offer you my project. I need assistance setting up Freepbx with Twilio SIP trunk to work properly. The system currently have two problem need to be solved. 1. Dial out plan setup - As of now, I need to input "+1" then the phone number to dial out successfully. I would like to eliminated the "+1" input. 2. Cannot call certain phone number - If I place outbound call from pbx to my cellphone, the call get drop immediately. If I placed a call to my office phone then I have no issue. My cellphone and office phone have two different area c...

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    Hi aqsyounas, I noticed your profile and would like to offer you my project. I need assistance setting up Freepbx with Twilio SIP trunk to work properly. The system currently have two problem need ...profile and would like to offer you my project. I need assistance setting up Freepbx with Twilio SIP trunk to work properly. The system currently have two problem need to be solved. 1. Dial out plan setup - As of now, I need to input "+1" then the phone number to dial out successfully. I would like to eliminated the "+1" input. 2. Cannot call certain phone number - If I place outbound call from pbx to my cellphone, the call get drop immediately. If I placed a call to my office phone then I have no issue. My cellphone and office phone have two different area co...

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    I need to configure a remote sip extension to incredible pbx on my raspberry

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    i need to configure a remote sip extension to incredible pbx on my raspberry.

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    Hi sanaakhlaq, I noticed your profile. I am the developer of a Multitenant PBX and I'd like to start Automatic Testing using Selenium IDE (if you think it will be the best tool to use). I am looking for a skilled person able to arrange all the tests, so I can just run them before any new release. I have used Selenium IDE for a little time, but if you think a different product will be best, let's discuss. There are more than 400 pages to be tested.

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    we need an asterisk developer with a2b and opensips experience. dont bid if you dont have this experience, as we need to fix a bug in a script.

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    Hi, i need someone very comfortable with 3CX IP PBX System, i have done by myself the configuration of Trunk and extension and some other things now i need to make my spa122 spa112 and gxv3275 working with it Thank you in advance

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    I would like help to integrate our asterisk PBX to Vtiger. Current module is not working

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    1) Develop a web-based softphone which can dial a single number or multiple numbers; 2) Developer(s) must have experience in VOIP/telecommincation industry, good understanding of Soft-switches, Interconnection, Mapping and Routing,...softphone which can dial a single number or multiple numbers; 2) Developer(s) must have experience in VOIP/telecommincation industry, good understanding of Soft-switches, Interconnection, Mapping and Routing, and solid programming experience in Asterisk. 3) Individual developers preferred. Company/team bidders must allow for direct communication with developer. 4) The bidder must be time-oriented, able to work and complete project on schedule. 5) Long-term cooperation opportunity for quality bidders. Note: Please don't bid if no experie...

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    hi i need to setup puppy linux that can run from usb + asterisk provisioing based on mac address , if you can help then let me know.

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    Need expert who can help us setup puppy linux that can run from usb and can pull asterisk profile on mac address base from a remote provisioning is a simple project ,kindly do not bid if you do not have any experience with puppy linux .

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    pbx Ended

    I am looking for someone to work on and set up my PBX it is a Goautodial/Vici dial program

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    We have already setup PBX using Elastix and we are planning to enable more features in our system. Such as Setting up Elastix in a local server, Call hunting, Outgoing call restriction for selected extensions, Implementing billing software, Call tracing, Call quality should be good, Call disconnections should not be the there, Documentation.

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