Softphone asterisk openfire jobs
Install vtiger connector to running Asterisk 11 and vtiger 7.2
This is to change a small software program into an ISO file for easy downloading by agents of a call center and make some adjustments to the existing software . Must use Teamviewer to connect to our Multi Disk Server.
We had to reinstall an asterisk server, and now the dahdi T1 cards are not refelcting that they are bing programmed for a T1. This work needs to be able to worked on now
i try to connect with asterisk throw pami but got some errors any help
Searching for somebody who can find out issues with asterisk and fix it. Thanks!
...and 1 day to deliver 2nd part please bid according to your time cost to complete the project. We will create initially milestone for the 1st part of the project. Once its concluded successfully we will create the 2nd milestone. 1st) Build, install and configure for simple tests the open source project Its a 1-2 hour task if your have experience with SIP and asterisk - Milestone Max is $100 and release will be on installation completion - time to deliver 1 day. Awarded freelancer should set this up on his server and we'll make a call to/from the telegram SIP gateway and if it works we will offer remote access to our server to do the installation on our server and instructions sent on how to install and configure. 2nd) Create a service like
Need configuration of the FreePBX to send SMS by gsm modem. My Freepbx works on raspberry pi 4 B with 16gb card, and has 1 modem connected with a sim card. There is 1 extension SIP trunk connected GSM to VOIP, and VOIP to GSM. Im connected to freepbx by zoiper app in my mobile phone. In short way: In my mobile phone im using "zoiper app" which is connected by SIP to Freepbx. I...send and receive sms from gsm sim card (modem is huawei e160 in raspberry with freepbx) using a zoiper or other voip app. Curently text mode in zopier doesn't allow me to send or receive gsm sms. [check images] All recived sms messages going to previously email address (spam inbox) in freepbx settings. IMPORTANT: Need skilled person to do the job, only serious performers and freelancers in a...
ASTERISK EXPERT to set up PBX We are looking for an Asterisk Expert to set up PBX. Interested freelancers can bid for the project.
Hi guys! I need an experimented guy with a heavy knowledge of Asterisk , SIP Protocol , PATTON E1 gateway, Issabel IPBX and RAD ASMI-52 modem for assistance on E1 trunking configuration between the PATTON gateway and my Issabel IPBX and E1 signalisation troubleshooting between my Patton and my ISP RAD ASMI-52 modem. Patton model is SN4171/1E15VHP.
Searching for an expert who can audit asterisk server and fix existing issues.
Hello, We are looking for the expert who can work on the Asterisk server and telephony integration.
We need help to install a Kamailio infrastructure with Asterisk, for few millions of calls per day. Solid experience required on Asterisk and Kamailio. Special Development Module for AS* and Kamailio. Development on the core code for AS*. Training. Database configuration. Monitoring script with Zabbix. GUI interface. Physical appointement in France (Paris-Toulouse) or Swiss (Geneva).
I have cdr from asterisk ( my sql db) I need to choose several columns ( filter them accurding to the info inside) Than to send excel sheet on spesific time to my email. I can add many reports as i want. It should be a tool for windows.
Create a service like We need: 1) Our clients calls from Telegram app to our Telegram account should go to our Asterisk PBX via SIP 2) Be able to from our Asterisk PBX call our clients Telegram accounts. 3) Handle 50 concurrent calls Open source example:P roject Here you can see how the bot works Libraries used: Current TG2SIP version can handle about 100 simultaneous active calls with opus@48k codec on ordinary hardware. 50% milestone on delivery of working project connected to our PBX for tests 50% milestone on delivery of source
Want to convert avaya 1608-l to sip so that it can be used on asterisk pbx
Create a service like option to use their open source project so that our clients can call our asterisk PBX from their telegram accounts. Here you can see how the bot works Libraries used: Supported call types are: SIP->TG call by username SIP->TG call by phone number (if callee is registered in telegram) TG->SIP to fixed SIP URI (can be set up in TG2SIP settings). Current TG2SIP version can handle about 100 simultaneous active calls with opus@48k codec on ordinary hardware. 50% milestone on delivery of working project connected
Searching for an expert who can audit asterisk configuration and can make the PBX secure from outside.
We are looking to develop an app (very much like UBER, but for other services), it must include the back-office website to administer all request and integrate to our current telco and VOIP network (Asterisk, and possibly Twillio for audio and video calls), appointment and project management platform. It must be able to track minutes, clients information, user information via a database.
we need asterisk configure in our local PC for telecalling and tele marketing purpose scope of work install & configure asterisk (essabella) basic setup trunk inbound and out bound voice clip sending moudule setup
Hi, we are a small startup from Poland. We are looking for someone who will help us with Asterisk integration. We have our own application based on Laravel Framework. We want to improve our app about dialer features most calling, receiving calls and recording calls. Do you have any experience of that type of work which could help us?
hi fix app issues , we are using Openfire and kotlin, You have to work on anydsk only.
need to enable call recording on asterisk service for 20 seconds to match available recording so the can take further step
Dear Sir or Madam, I am searching for an expert in VoIP and Asterisk or alternative program. What ist he problem/task? I am going to create an android app like „Marcophono“, which you can find in the german playstore. With this app Users can do prank calls. It works like this: 1)User A choose a scenario in the App. 2)In this scenario User A can see different Audio Messages. 3)User A calls User B with the App. 4)If user B answer the call, User A can press different buttons in the App like „Hello my friend“, „How are you“ etc. User B hear that messages directly live, but not the voice from User A. Important: User B did not have this app installed on his phone. Thats the point I need your experience. I think the owner of „Marcophono“ did ...
i need a php script which connects to ami asterisk (this is already done) then it needs to set dynamic agents to be remove or added to/from a specific queue and also we need a shell script which logs off all agents from a specific queue
hi fix app messaging issue , we are using Openfire and kotlin
i need a php script which connects to ami asterisk (this is already done) then it needs to set dynamic agents to be remove or added to a specific queue then we need some jquery tool which polls from ami and displayes if user is logged in to queue or logged off. and also we need a shell script which logs off all agents from a specific queue
Hi vladimirumnov, I noticed your profile and would like to offer you my project. We can discuss any details over chat. Trunk capable of making 20 calls on softphone, but limited to 2 calls on FPBX.
hi fix app messaging issue , we are using Openfire and kotlin
hi fix app messaging issue , we are using Openfire and kotlin , work on anydesk only
We are looking for some Asterisk work on a number of items. We are not using freepbx or other GUI. The current tasks consist of the following: The configuration is two SIP trunks from Twillio. One trunk works the other does not. Need to troubleshoot. All calls coming into the default extension of either of the trunks must simul ring to all extensions existing now and any that are added in the future. On one trunk this is configured for existing extensions only. Needs to work on newly added extensions as well as second trunk. Notification of all missed calls to default extension must be sent to a specified mobile number via text. Any voice mail message left to that extension must also be sent via text and email. Notification of all missed calls to other extensions must be s...
Task - Enable DTMF ( press 1 to connect) if pressed then forward to an external number and if no response disconnect the call.
Asterisk Freepbx Admin - Zoho phone bridge Integration
android messaging stopped working - openfire
Hi Ram i have two asterisk conmected via IAX, when call is NO ANSWER or BUSY or CANCEL im not getting good Sip cause but 503, exemple client send call he get ringing 180 than after timeout (no answer) he get 503 instead of 408, the gateway is sending correct sip cause to 1st asterisk but always congestion(503) except when call Answered I think something wrong in or or in my My Architecture is CLIENT --SIP--> ASTERISK --IAX--> ASTERISK --SIP--> GW
Ura com reconhecimento de voz IA para asterisk. Text-to-speech e reconhecimento de voz.
When making a outbound call from Skype for Business softphone unable to hear the ringing on SFB app. But the call rings at the other end and when answer the call works fine. Current setup is when calling from SFB has a trunk connecting to Asterisk freepbx and the SIP trunk to the provider is configured in Asterisk. it would be great to have someone who has knowledge in both Asterisk and SFB server.
Interconnection between WINDOWS SOFTSWITCH that has PUBLIC IP, and Asterisk PBX (DYNAMIC IP),, We want to connect the Asterisk PBX as a trunk to the windows softswitch so calls from Softswitch to Asterisk PBX can be successfully connected.. Please note: DYNAMIC DNS is not preferred.. Thank you.
I need to register my ASterisk PBX to my Softiswtch,, my asterisk PBX has dynamic IP
Asterisk Wallboard. Code php 5.6 Same layout as image below Management portal to control Wallboard Queues to show and which Agents by Queue to show and other settings as colour etc Able to setup more than one wallboard. Able to assign colours to Queues or Queue Groups Lookup Pause name and Agent name from another MySQL DB Only show top queue box if there is a Call waiting Colour change on Queue Box if over 5 calls or 10 calls waiting. (Yellow, Red) Colour change on Agent if over 20 minutes talk time and 40 minutes talk time. Allocate sort order to Queues or Queue groups. Agent last call record is the last call from any Queue
i want to change the path of my extension file in asterisk
I have a call center software I need a vici dial asterisk server expert who can change my system layout & make some systems to work properly according to my requirements
Hi, we are looking anyone who can create web interface for asterisk where we done below task 1. add / remove sip trunk 2. check sip trunk status 3. realtime calling (live calls) 4. force/terminate calls 5. generate calls from mysql and send CDR to any Mysql 6. generate TTS calls from mysql and send CDR to mysql 7. Capture User Input (DTMF) and send to mysql along with CDR thanks
Asterisk, PHP, agi, xml call Task description We need information from a web service, from an XML Soap Call. We need to be able to call this php-agi script from asterisk, passing it variables such as telephone numbers or account numbers . XML Calls (SOAP) There are three calls from XML that we would like to call. These are real links 3.PayByCreditCard_New
Hi, I m looking for asterisk devoloper for api integration. We have a IPPBX based upon asterisk and we want IF customer INPUT through DTMF, It should read balance and all.
Install and hook up VOIP phone server (asterisk) Users are recognized by registered phone numbers in profile. Ability to record product name & price (2 separate fields) for each product. Main menu: - Sale menu: leaf through all available products for current sale, select products and add {quantity} to cart. - Existing order: leaf through all items in cart one by one (Press x to , with quantity and price (per item). Ability to edit item order (add / reduce quantity, remove from cart). - Automatically leaf through all order items in sequence. (no editing) - Hear order status message (pending payment, processed, shipped etc.) Previous experience is a must.
