Asterisk dialplan jobs
I want a special setup on our asterisk server. If an agent dials an agreed extension, then the asterisk should automatically dial an agreed number as you can see below. Monday between 8: 00-16: 00 Danish time, then the call should ring between the following two setups. The first time it is dialed, it must dial 004580808080, after 15 seconds, then it must interrupt and then dial up to 004580808081, if no answer is answered again within 15 seconds, it will dial the last number 004580808082. the answer does not interrupt the call so the agent can talk to the customer again. The second time it calls, it must call 004580808081, after 15 seconds, then it must interrupt and then call up to 004580808080, if no answer is answered again within 15 seconds, it will call the last numb...
Hello can you install Asterisk GUI in Ubuntu ? need your assistance to configure and . If you can please message me urgently and ASAP.. Thanks.
Hello can you install ASterisk GUI in Ubuntu ? need your assistance for configuring
Hello, I need your assistance for Asterisk GUI installation.. my server is installed but we need GUI configure..
Hello Vasile, I need your assistance to configure my Asterisk server GUI.
All is installed, asterisk,npm,node js, google cloud you need to integrate speech to text using the v2 key from google
I just need a sip PROXY that will allow me to receive traffic from my customer a USA based telecom company (IP Based authentication) and then send the calls to my vendor who uses an Asterisk and will authenticate my IP. Capacity for 60 simultaneous calls, no rtp relay needed, just signaling handling. Need to be able to download CDRs in csv. file extension. For this project you need to indicate server specifications, I will use services. Also as an additional feature, I will a simple and plain php page showing all active live calls with its current duration.
build the Callcenter via Asterisk server up to 30 extensions welcome prompt Database for call center information IVR and SMS based (give the Details Via chat) technical Integration with Telecuminication Shortcode Routing call transferring / call record / billing record/ DND/ etc few more feature should be mention via chat. looking for an active and responsive guy not only give the details and then he or she busy with other works.
Asterisk with WebRTC for Inbound call center. This is only for freelancers with a very broad knowledge-base and who know how to get things done with the terminal. Some requirements: FreePBX/Asterisk w. WebRTC, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. There are a few fixes to make and the system must be checked to get it ready for production. The physical Servers are installed and most of the software is also setup. A late Ubuntu OS and Teamviewer is required to login. --- Project will be divided into milestones, to be discussed.
Hello. We need someone who can help us to configure freepbx and asterisk.
Hi Anton, you assisted us with FreePBX setup using AWS singapore last year. I rebuilt the instance in AWS Hongkong - I wasn't able to use the image due to compatibility issue of the network adaptor. Anyway the new instance will freeze (asterisk) every 1 to 2 weeks, and I m not sure why. by any chance you can help us looking into this and resolve the issue? Best, Johnny
Hello, I need help with GoAutoDial 3.3 configuration. I'm from Poland and i want to call from polish voip operator () to other polish numbers. The standard polish phone number contains just 9 digits +48XXXXXXXXX. I can set up my own lists and users i just need it to dial out. The GoAutoDial is self-hosted on company server. I just need working dialplan but for polish numbers. Example dialplan is not working: exten => _48X.,1,AGI(agi://) exten => _48X.,2,Dial(SIP/@{EXTEN},,tTo) exten => _48X.,3,Hangup() ASAP
Important - no PBX / Asterisk / SIP -> only XMPP CALLING We are looking for a developer who can develop a plugin for Android app to CALL (not push to talk broadcasts) but truly calling via XMPP. When needed also a plugin for Jabber server to enable XMPP CALLS. ONE-ON-ONE / END-TO-END XMPP CALLS between two users as xmpp1@ - xmpp2@ without the use of a Asterisk / PBX server - truly only web based XMPP CALLS. Budget $1000 for the ones who truly have experience in XMPP CALLS.
Hello, We are looking for a developer who can develop a plugin for Android app to CALL (not push to talk broadcasts) but truly calling via XMPP. When needed also a plugin for Jabber server to enable XMPP CALLS. ONE-ON-ONE / END-TO-END XMPP CALLS between two users as xmpp1@ - xmpp2@ without the use of a Asterisk / PBX server - truly only web based XMPP CALLS. Budget $1000 for the ones who truly have experience in XMPP CALLS.
I have extensions which are forwarded to external numbers. I need a method which will allow me to listen to one of these forwarded calls. I will be listening from an inside extension. we'll be listening to a call that came from the outside that was forwarded again back to the outside. I can currently listen to a call before it is forwarded but my current method does not work once the call is forwarded.
Dear all, I am looking for someone who worked closely with FusionPBX in deployment and configuration better yet customization. I have a good experience in Asterisk and Issabel, and for the same of multi-tenant feature which is not available on Issabel, I am intending to move to FusionPBX I am expecting someone who can give me a tutoring session to walk me thru on how to best utilize and familarize myself to use FusionPBX I know that most of the features are in common between Issabel and FusionPBX but the options are scattered over. the best tutoring session would be covering the 5 sections: 1- Home 2- Accounts 3- Dial Plans 4- Applications 5- Advanced on
Dear all, I am looking for someone who worked closely with FusionPBX in deployment and configuration better yet customization. I have a good experience in Asterisk and Issabel, and for the same of multi-tenant feature which is not available on Issabel, I am intending to move to FusionPBX I am expecting someone who can give me a tutoring session to walk me thru on how to best utilize and familarize myself to use FusionPBX I know that most of the features are in common between Issabel and FusionPBX but the options are scattered over. the best tutoring session would be covering the 5 sections: 1- Home 2- Accounts 3- Dial Plans 4- Applications 5- Advanced on
This is for an inbound call center with Asterisk-FreePBX-WebRTC. It still needs some fixes. First, incoming calls must be transferred to internal extensions and to outside land lines. Second, the PBX must call agents who forgot to register for their shift. This system works with WebRTC, so only people who know this subject with it's implications, should apply !
We are a company with 15 sites in Portugal. with over 40 employees spread across the various centers. We intend to have a solution to talk between centers via IP, conference calls, IVR messages, automatic call forwarding, etc.
The project involves migration of existing Asterisk PBX to a Azure cloud. There is existing Azure account. Install FreePBX on Azure cloud. Install Linux and FreePBX GUI. After the installation, migrate exiting asterisk configuration and voice prompts and confirm that the system works accordingly. Copies of the Asterisk files will be provided (have to be specified). I would also like consultation of the benefits and pitfalls of hosting the FreePBX in the cloud. I also would like other alternatives for hosting and the prices that are accrued for each SIP connection and call that is made. If there is a more efficient and inexpensive way to host online please advise.
Details will be shared winning bidder. please bid if you have the experience.
To make assisted transfers on FreePBX/Asterisk with one click. FreePBX / Asterisk with WebRTC ! This will be the first milestone to be discussed. For the whole projrct, see attachment.
...of Vicidial and it should work with Asterisk 1.6 or 1.8 or both. (or higher) All Vicidial UI and branding has to be replaced with our custom brand name, and interface should be changes sufficiently to look different. Logo branding removed Marketing automation Lead analytics dashboard Analytics dashboards SMS +++ To Email and Phone WhatsApp through the system Inbound Call Tracking Outbound Call Tracking Call Analytics Live chat Further the software needs to have some 30 additional reports/metrics displayed in addition to the standard reports available on vicidial. The code needs to be encrypted and should be packaged as an ISO.******* and also wget command format.********* The final mod vicidial should be able to provide per users/agent licensing Asterisk and Vicidial ex...
i want make rules in my server 1- i want port number 1 call number 2 to i mint 2 i want when i send call for port number one when it finsh it make stop this port 2 or 3 min exe exten => _[*#0-9]!,n,Dial(Mobile/A0/${EXTEN},60,tTm,K) exten => _[*#0-9]!,n,Dial(Mobile/A1/${EXTEN},60,tTm,K) --------------- A0 port number 1 A1 port number 2 If a user make a call from a sip account, asterisk will send that call out using any one of the port from A1 to A16. If A15 was selected for that call, then after that call ends, A15 will not be used for 2 minutes for sending any other calls User Avatar and also new call from a15 should be dialed to any other 3 ports one by one for 30 seconds each so if we wait for 2 minutes and then call other port for 30 seconds, then other for 3...
i want make rules in my server 1- i want port number 1 call number 2 to i mint 2 i want when i send call for port number one when it finsh it make stop this port 2 or 3 min exe exten => _[*#0-9]!,n,Dial(Mobile/A0/${EXTEN},60,tTm,K) exten => _[*#0-9]!,n,Dial(Mobile/A1/${EXTEN},60,tTm,K) --------------- A0 port number 1 A1 port number 2 If a user make a call from a sip account, asterisk will send that call out using any one of the port from A1 to A16. If A15 was selected for that call, then after that call ends, A15 will not be used for 2 minutes for sending any other calls User Avatar and also new call from a15 should be dialed to any other 3 ports one by one for 30 seconds each so if we wait for 2 minutes and then call other port for 30 seconds, then other for 3...
PLEASE READ THIS FIRST ! Asterisk with WebRTC for Inbound call center. This is only for freelancers with a very broad knowledge-base and who know how to get things done with the terminal. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. There are a few fixes to make (see attachment). The physical Servers are installed and most of the software is also setup. A late Ubuntu OS and Teamviewer is required. --- Project will be divided into milestones, to be discussed.
High level description: The IVR need to "collect" digits from the coller (personal ID number) via DTMF or Google voice to text service. The collected info needs to be stored on SQL table and passed as a parameter to dynamic (user defined) URL Task prerequisite: Asterisk 11, freepbx 2.11 Asterisk 16, freepbx 15 Full installation doc. All parameters will be via php page on /var/www/html/over Full error log will be stored in dedicated SQL tabel. IVR- caller identification settings IVR- caller identification settings page should include the following fields: Queues dropdown list- caller identification will be enabled in the chosen queue. Identification (checkbox)- user will choose which identification option to enable: Dtmf Voice Voice and dtmf Dtmf l...
sip asterisk specialist to get in touch with our team in order to drive inner development of a core product. SIP webrtc, SIP API sip asterisk master configuration
PLEASE READ THIS FIRST ! Asterisk with WebRTC - A real Specialist is needed.! For Inbound call center with Asterisk and WebRTC. This is only for freelancers with a very broad knowledge-base and who know how to get things done with the terminal. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and most of the software is also setup. There are a few fixes to make. First: - Assisted Transfers to agent's extensions and to PSTN lines, calls can be taken in random order and Caller must not hear himself (echo). Ubuntu and Teamviewer is required. --- Project will be divided into milestones, to be discussed.
Asterisk with WebRTC Specialist needed.! PLEASE READ THIS FIRST ! For Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and most of the software is setup. There are a few fixes to make, like Assisted Transfers to agents and PSTN lines, calls can be taken in random order, Caller must not hear himself (echo), PBX calls agent who didn't register, No-store cache control for Apache, etc. Teamviewer is required. --- Project will be divided into milestones to be discussed.
Hello, I am looking for a data scraper who can get me leads from America from Instagram for the personal trainer niche. I need 1,000 leads for now. If you do a good job, I will hire you for a lot more business. I need the following from each personal trainer / health coach’s Instagram: First name*, last name, business name, phone number* email address*, City*, State*. Anything in the asterisk is mandatory for you to fill out. If you cannot fill out, don’t add that lead, go to the next. Please let me know what your price is. Thanks,
Asterisk with WebRTC Specialist needed.! PLEASE READ THIS FIRST ! For Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and most of the software is setup. There are a few fixes to make, like Assisted Transfers to agents and PSTN lines, calls can be taken in random order, Caller must not hear himself (echo), PBX calls agent who didn't register, No-store cache control for Apache, etc. Teamviewer is required. --- Project will be divided into milestones to be discussed.
I want simple dialpad with all dial functionality in web application (html as ui) and call flow should be initiated from asterisk / pbx
Our company is in the process of moving many of our services. We would like to have a Asterisk VOIP Server installed and configured on a new AWS Instance under our AWS account. Candidate must have experience with Asterisk.
Looking for a developer with experience working with Asterisk and FreePBX. Looking for a billing solution.
New enhancements on asterisk dialplan I have 2 termination partner. Company A and B. They have their own DIDs to mask. They will provide their DIDs and we will store them in dids table to mask original calling ID. Call will come asterisk will choose one DID from did table and send the call to termination partner. it can be "company a" or "company b" but if it chooses company a DID it should send it to company a trunk. If a previous call is forward to certain DID then repeated calls will use DIDs from same company (instead of remaining limited to only that DID)
We will provide you with a clean install of Alpine Linux running on a raspberry pi. You install and configure the following software. hylafax+, iaxmodem, asterisk, freepbx We will provide you with port 22 ssh access to this device, we can also provide additional access if you require it.
JobDiva VoIP Integration Project Details, 1- Click to Dial: JobDiva supports Click to Dial using HTTP call or RESTFul API. Please check with the VOIP provider for the method that is supported and let us know the details. 2- Incoming Call Screen Pop: two methods are supported. a. Use the JobDiva API web services’ searchCandidateProfile and/or searchContact method to locate a matching candidate or contact. Pass digits only (without leading 1) as phone number parameter. b. Call a JobDiva page to get a display the search results Pass the phone digits only (without leading 1) as the ANI parameter. Check with Jobdiva to use either or www2.jobdiva.com. 3- Note for Incoming or Outgoing calls: Use the JobDiva API web services’s createCandidateNote and/or createContactNote met...
PLEASE READ THIS FIRST ! -- Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical setup is done and most of the software is installed. There are a few fixes to make, like Assisted Transfers, calls can be taken in random order, Caller must not hear himself, PBX calls agent who didn't register, No-store cache control for Apache, etc. Teamviewer is required. --- Project will be divided into milestones.
Hi Asterisk O., I noticed your profile and would like to offer you my project. We can discuss any details over chat.
i want make rules in my server 1- i want port number 1 call number 2 to i mint 2 i want when i send call for port number one when it finsh it make stop this port 2 or 3 min exe exten => _[*#0-9]!,n,Dial(Mobile/A0/${EXTEN},60,tTm,K) exten => _[*#0-9]!,n,Dial(Mobile/A1/${EXTEN},60,tTm,K) --------------- A0 port number 1 A1 port number 2 If a user make a call from a sip account, asterisk will send that call out using any one of the port from A1 to A16. If A15 was selected for that call, then after that call ends, A15 will not be used for 2 minutes for sending any other calls User Avatar and also new call from a15 should be dialed to any other 3 ports one by one for 30 seconds each so if we wait for 2 minutes and then call other port for 30 seconds, then other for 3...
VICI / PHP fullstack developer with some experience in designing and integrating front-ends. We are looking for a developer+designer with some experience with VICI (doesn’t need to know much about asterisk but some knowledge of it would be good). This position will contribute to a creating, and maintaining a custom designed interface for our VICI setup (primarily the agent screen design to start with, following up with the admin theme design and implementation), which would be used by our agents. Apart from working on VICI, the developer will also be working on some other in-house projects. Responsibilities: • Writing clean, secure and quality code that scales with concurrency. • Hands-on development, debugging and on-going maintenance of new and existing code (PHP, ...
Hi Asterisk O., I noticed your profile and would like to offer you my project. We can discuss any details over chat.
Please call me for asterisk or vicidial related projects. Also if you need leads for cold calling, I can provide you numbers based on regions and operators. All genuine numbers.(No customer details)
A module where the caller has an option to exit the queue without losing their position and a callback occur when there position is up next on the queue.
i need a guide about how to implement it: also how to send a php post to originate it in order to use in some projects
lease if you haven't skill with Asterisk and linux don't offer me i install freepbx on raspberry when i make call its ring in end point but no odio in both side see log -- Added contact 'sip:200@;rinstance=5D817960' to AOR '200' with expiration of 600 seconds == Contact 200/sip:200@;rinstance=5D817960 has been created -- Removed contact 'sip:200@;ob' from AOR '200' due to remove_existing == Contact 200/sip:200@;ob has been deleted -- Contact 200/sip:200@;rinstance=5D817960 is now Reachable. RTT: 562.179 msec == Setting global variable 'SIPDOMAIN' to '' == Using SIP RTP Audio TOS bits 184 == Using SIP RTP Audio TOS bits 184 in TCLASS field. == Using SIP RTP Audio
PLEASE READ THIS FIRST ! You must use a PC with a late Ubuntu OS. -- Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, FreePBX/Asterisk, SSL, Ubuntu, etc. The physical setup is done and most of the software is installed. There are a few upgrades and fixes to make. Teamviewer is needed. --- Project will be divided into milestones.
