I have been using my own class 4 softswitch for about 3 years now but decided to change to a rented voipswitch. Unfirtunately am struggling to understand it fully. I need help with configuring expecially, getting the pbx module to work. Need to understand fully how voipswitch pbx works and get it working.
Hello geeks, we need Integrate Asterisk HubSpot and Asterisk Laravel custom CRM, I`m ready discuss project derails feel free contact me , Best Regards GS
I need an end user interface to manage a control basic features on Asterisk using FreePBX. The user should be able to create ring groups, control inbound routes, control call forwarding, control DIDs, view CDRs and check voice mails.
Build a Reporting System , based on Vicidial database architeture, (call_log) for the calls made outside Vicidial Crm (so manual calls) , and (carrier_log) to see if actually the call went through They get stored in call_log table with structure as in () attached The UI should be based in AdminLTE The report should be similar to vicidial reports that ...jpg) attached The UI should be based in AdminLTE The report should be similar to vicidial reports that use Perl, to minimize database overload , the report structure should be as in (screenshot_5) , The db_connect and language file should be separate , There is the need to alter the call_log to store the sip result too No bid if you dont have previous experience with vicidial and asterisk Thanks
main iso link https://www.freelancer.com/users/l.php?url=http%3A%2F%2Fdistro.ibiblio.org%2Fpub%2Flinux%2Fdistributions%2Fpuppylinux%2Fpuppy-5.2.8%2Flupu-528.005.iso&sig=f8bb0783ca9a793504be0860f0aa7c237556055768cc030588618ea121ca50d6 you have to install fastd asterisk openvpn teamview 7 Google Chrome thats it if you can please bid i will boot usb if u did not made before or u dont know how to made iso please dont bid price FIXED under $100
Hi bro, I want to configure, remote VOIP over VPN, I have on my local network "MyPBX" router, WRT router (OpenVPN client), I also have a VPS Centos7 (OpenVPN Server), I want people from the internet to make calls using my PSTN lines throw "MyPBX" router, I want to implement two things: 1- lin...VOIP over VPN, I have on my local network "MyPBX" router, WRT router (OpenVPN client), I also have a VPS Centos7 (OpenVPN Server), I want people from the internet to make calls using my PSTN lines throw "MyPBX" router, I want to implement two things: 1- link remote branch VoIP phone to call extensions in HQ and to use local PSTN in HQ, 2- any one can use sip softphone to link to the HQ PBX, and to call extensions in HQ and to use local...
Hi kingAsterisk, I noticed your profile and would like to offer you my project. We can discuss any details over chat. The system make calls and leave telephone message, reply to the question automatically. Analyse total number of phone calls you made. The Asterisk FreePBX or PBX solution, they don’t have the same function. (1) To build up a list phone calls you made by Excel or CSV (2) To upload voice (3) To create a campaign In addition, another function: IVR schedule The specified function is transfer all incoming calls and transfer all calls to the preset telephone number. When the customer ring up and push the keypad which shows his required. Telephone will response to a transfer calls to the telephone number which one is best for cust...
Hi Ivan V., I noticed your profile and would like to offer you my project. I have a zycoo coovox u50 ip pbx. need to install and get up and running with Zycoo UC Pro. We can discuss any details over chat.
Hi Ambiorix R., I noticed your profile and would like to offer you my project. I have a zycoo coovox u50 ip pbx. need to install and get up and running with Zycoo UC Pro We can discuss any details over chat.
IVR Requirements Install odoo-asterisk connector in odoo backport modules as necessary from odoo 10 to odoo 9 configure asterisk for inter-operability with odoo Install and configure webrtc via odoo and asterisk install available odoo webrtc module configure webrtc module to route through asterisk configure asterisk to accept and route webrtc Configure auto-dialing groups in odoo using contact, leads, or any other editable/custom group. Add calling, sms, multimedia and voice broadcasts to odoo follow up options Optimize server configuration for reliable high audio and video quality configure settings for sip, users, codecs, etc for best quality configure PSTN connection(s) for best quality Enable chat via soft-phone Configure asterisk/odoo ...
Требуются очень хорошие знания по Астериск. Задачи: Создать концептуальную модель следующей системы: -получение номера телефона звонящего абонента (звонок идет на zoiper, а номер нужно показывать в веб интерфейсе приложения открытого на ном же компютере), автоматически поднимать трубку на Зоипер при нажатии на кнопку поднятия трубки в приложении (через интеграцию астериск) , логин в астериске/зопере во время логина в систему, настройка очередей обзвона через приложение
Integrar o pipedrive por API com asterisk PBX para fazer Ligações via pipedrive registrar status das chamadas agendar atividade de ligação quando chamada mau sucedida
I need an Asterisk expert with some experience . There are 3 phases : Planning , Install and configure then Maintenance for 6 months. I want this to be an hourly and fixed price job both that can be agreed upon .
we need a skilled team/idividual to help to port this module from Odoo 10 to 9 and extend the features to truly bring the best of asterisk, kamelio, webrtc and multimedia call center solutions
Hello, I have an FXO gw SPA400 behind a firewall /home router. It has a local IP : 192.168.1.x. The asterisk server has a public IP address. I have forwarded UDP ports 5060 and 10000-20000 in the router to SPA400 IP address. The SPA400 can register to asterisk server. The problem is that: 1) huge delay (more than 20 sec) for outgoing PSTN calls. There is no big delay for incoming calls. 2) PSTN called/caller are unable to hear anything apart from the ring Please apply only if you have worked on similar setup before. The project will be considered as successful when both issues are resolved. Thank you.
Hello, Our Asterisk PBX application went down on our vps. When it came back up we got this error message: "action could not connect: can't connect to local mysql server through socket '/var/lib/mysql/(2)" ONLY experienced Asterisk techs should respond, thank you.
i need to create an ajax api call from my TLD domain here also the "info" part needs to act like an asterisk subdomain without a wildcard so when you refresh the "info" page for example https://sim.psim.world/showdown/info=?12345678910 it changes regardless of where you search like https://sim.psim.world/showdown/info=?46589645764625
Seeking a senior level developer with deep experience building custom IVR systems for customer support and open ticket management. . - installation, configuration, and maintenance PBXFREE / Asterisk based solutions - integration VoIP systems (Asterisk) and CRM - developing IVR and application (incl. AMI/AGI based) for Asterisk - Twilio integration experience The main idea is to identify our customers by caller id, check to see if they have open tickets and connect them to proper agents or resolutions using speech recognition, keyed responses and or knowledge base. Attached is a more detailed description of how the IVR should be setup with our current CRM. We are currently using FREEPBX but its very limited. Please describe your experience building custom t...
I need to create a branding proposal for IT Voip Solutions and AC Global Tech Solutions. The type of service are: IT Service, IP Telephony, IP PBX, GPS Services and Online Backup Data. For that, we need to create brochures and flyers in english and spanich.
Build a Reporting System , based on Vicidial database architeture, (call_log) for the calls made outside Vicidial Crm (so manual calls) , and (carrier_log) to see if actually the call went through They get stored in call_log table with structure as in () attached The UI should be based in AdminLTE The report should be similar to vicidial reports that ...jpg) attached The UI should be based in AdminLTE The report should be similar to vicidial reports that use Perl, to minimize database overload , the report structure should be as in (screenshot_5) , The db_connect and language file should be separate , There is the need to alter the call_log to store the sip result too No bid if you dont have previous experience with vicidial and asterisk Thanks
hi, we need asterisk support. only experienced person can apply. thanks.
We have a Chinese made HSUPA dongle, which has voice, sms, internet and storage features.... it is not connecting to local service provider via sim. we need it to the our local gsm network so all features can be used in asterisk... we believe it may be suck in WCDMA mode... we just need the sim to connect to network in GSM mode and be able to run multpl edongles on the same system...
Integração entre o sistema de telefonia Asterisk com uma base de ERP para pesquisa de dados de usuarios
need a click to dial application for windows that uses AMI examples below this has to work in programs and web browsers which is why i'd rather it be a windows program the obviously configuration variables must be editable
Hello, I have an active project for asterisk. Would like to take a look if you interest on it ?
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.
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
I want a VOIP PBX with a capacity of six extensions and call forwarding and also generate DIDs . Ideally knowledge of XML & VOIP or superior skills will be preferable.
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 e...
I want to build a VoIP telecom company. We spoke yesterday. I think asterisk is the best for the job. all the properties of the Asterisk server to be managed via the panel I want. like for example the combination of a2billing and FreePBX. How do we work ? The long-term mean is constant if I want to work with a person who knows. Please tell me what you can do. For example, a snapshot of the database as 2-sided so redundant (cluster) to work as I want. for example, I want to place the server into the 5 continents of the world. These servers are geographically closest to the customer and connect it with the SRV record to a single domain for voice services provided over the server I want to be. The whole structure have to be redundant. Very good attentive visual theme you need to...
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 ...
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@0.0.0.0) - No matching endpoint found Also extensions are not ...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@0.0.0.0) - No matching endpoint found Also extensions are not showing connected or the IPs when showin...
...recovery procedures and disaster recovery plans. #Key Responsibilities: -Monitoring and analyzing existing services; -Resolving technical problems; -Automating IT processes by writing bash / php / sql scripts. #Skills&Qualifications: -Good knowledge on server administration on Linux (Redhat, Ubuntu, Centos) LAMP; -Understanding of VPN (OpenVPN,IPSEC); -Good knowledge of Nginx, Apache, MySQL,Asterisk; -Ability to write scripts(bash); -Understanding of Nagios, Zabbix and etc. #Must-have skills -Maximum attention to detail and analytical thinking -Orientation on research of additional information independently -Ability to organize workload taking full responsibility for what you do -Ability to provide information clearly and shortly -To be friendly and communicative #Work co...
...backup and recovery procedures and disaster recovery plans. Key Responsibilities: Monitoring and analyzing existing services; Resolving technical problems; Automating IT processes by writing bash / php / sql scripts. Skills&Qualifications: Good knowledge on server administration on Linux (Ubuntu, Centos) LAMP; Understanding of VPN (OpenVPN,IPSEC); Good knowledge of Nginx, Apache, MySQL,Asterisk; Ability to write scripts(bash); Understanding of Nagios, Zabbix and etc. Must-have skills Maximum attention to detail and analytical thinking Orientation on research of additional information independently Ability to organize workload taking full responsibility for what you do Ability to provide information clearly and shortly To be friendly and communicative ...
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.
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.
Looking to have a function created that will send a ringless voicemail to a mobile number. The function would accept a T...accept a TO phone number and an MP3 file (this is pre-recorded message we want dropped on users voicemail). The phone must NOT ring at all, no missed calls. We have a Twilio account but if its not possible through Twilio please suggest other means. We need to keep our costs down in what we are charged to send through which ever provider is used here. We are willing to setup a server (if Open PBX is a possible part of the solution) or whatever. The most important part here is the MP3 must go directly to voicemail. Prefer to have code in .NET, but if you can get it done correctly we are willing to use any code base. We can then add our own wrapper around ...
Add perspective to a vector graphic element (logo size) from a .eps file. Return file as .eps or AI and also as transparent png in 600x600. I attach the JPG of the background image we will use and you can see where the flattened circle is that all of the lines come from. I need the asterisk to be flattened in perspective to fit that circle. I DO NOT need the new asterisk logo added to the background image. It is actually just going to be in a Keynote Presentation as a separate object.
I nee...simple php script that will be in a folder inside the same asterisk server. Lets call it and it should be used as: As seen the php receives as parameter what it needs to make the call using specific extension. We will use this way: from our site we will make a link to the php which is in the asterisk server. The voice must go both ways with no other need to do, only adding the link with the correct parameter. We also will use this application from any browser typing the url with its parameters (the same aswe did in the link we explain when used from a site) The recording is not needed here as long as it could maintain the recording features of asterisk. Need a simple way to make installation and need code.
Need to make changes to extensions and multi-ring settings on various extensions for very old system with polycomm phones
Administrador de Telefonía VOIP experiencia en Asterisk comprobable, Linux , programación LAMP.
We need someone to help us maintain several Nortel and Avaya PBX sites around the world. We are hoping to find someone to cover the hours of 11 PM to 7 AM United States central time.
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
...supporting information systems. Installs new software releases, patches, updates and managers the overall configuration to ensure it is up to date and secure. Develops backup and recovery procedures and disaster recovery plans. Skills & Qualifications: Good knowledge on server administration on Linux (Ubuntu, Centos) LAMP; Understanding of VPN (OpenVPN,IPSEC); Good knowledge of Nginx, Apache, MySQL,Asterisk; Ability to write scripts(bash); Understanding of Nagios, Zabbix and etc. Must-have skills Maximum attention to detail and analytical thinking Orientation on research of additional information independently Ability to organize workload taking full responsibility for what you do Ability to provide information clearly and shortly To be friendly and communicative Salary ex...
I nee...simple php script that will be in a folder inside the same asterisk server. Lets call it and it should be used as: As seen the php receives as parameter what it needs to make the call using specific extension. We will use this way: from our site we will make a link to the php which is in the asterisk server. The voice must go both ways with no other need to do, only adding the link with the correct parameter. We also will use this application from any browser typing the url with its parameters (the same aswe did in the link we explain when used from a site) The recording is not needed here as long as it could maintain the recording features of asterisk. Need a simple way to make installation and need code.
hi, we need asterisk support. only experienced person can apply. thanks.
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.
Need to configure FXO/FXS gw (SPA3000) with asterisk.
ONLY PBX EXPERTS PLZ install and configure freepbx sangoma system and fix issues with existing installations ONLY PBX EXPERTS PLZ
