Asterisk pbx sugarcrmprojekty

...release the call and continue the transfer - x202 still sees "201" as Caller ID, until he picks up the call. Once he picks up the call, 222-222-3333 appears on the CID. I want to change the way the last step in the above scenario works. Once x201 presses Transfer again to release the call and continue the transfer, x202 should see 222-222-3333 on his CID display. This worked for us when we used asterisk. In , setting connected_line_method=update made this work correctly. If anyone has the right experience, I will be happy to hire you for this project. I do expect that you already have the experience and not wait to do the research after you are given the job. If you already know how to get this done, I don't expect to spend more than an hour with you working on ...

Currently we have several PBXs in Asterisk and there is a problem of Flooding of port 8089 Webrtc, it is not an attack since the traffic is valid and comes from IPs of our clients, this problem happens when the person who is using the Webphone has intermittence of internet, generating multiple connections in closing state that are observed in the Kernel log. We think that to solve this is to use an Opensips such as Webrtc Gateway and that the flooding is controlled from this point. We need a professional to help us install and configure an Opensips as a mid-registar WSS, which allows us to log the extensions found in Asterisk of the SIP type. Observation: The asterisk actualy is not Real Time, and for local devolpment with cannot change that.

Need to Confgure Goautodial V4 for a press 1 survey campaign and a IVR Blaster Campaign. need to configure dynamic login to self authenticate and stop sip attacks on the server

I m working in voip in Morocco from 2014 I have a system of asterisk Isabel the problem is when my client send me trafic give him an erreur 486 must be 503 In this case my client lost his traffic https://www.freelancer.com/users/l.php?url=https:%2F%%2Fusers%%3Furl%3Dhttp:%252F%%252Fpt%252FA42hBUju%26sig%3D50c67e1b3a7f27f20771fd2ee320f07b9eb302904f9e19edea3a28d328aafbdf&sig=dbffe148af4b5367f232a97a6acda81aea4364dfabcb09ccf07b413f3f1c3be9

Need a complete SKIN customization of opensource microsip. need also a little button that send information about operator status directly to asterisk pjsip server. will send details about image during chat.

1. SugarCRM 3. IONIC

We are currently Voip wholesale provider. But we are getting started with PBX Business . How can we set a service like this ? How can we create numbers and dial them to ring device in Kolmisoft

We are currently Voip wholesale provider. But we are getting started with PBX Business . How can we set a service like this ?

I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: You will probably also need to check the G729 codec configuration

PBX setup on server with any open source software for which I don't need to pay anything ps. which one want it on freepbx, don't come here, i have specially problems with freepbx

I want a PBX setup on server with any open source software which I don't have to pay

I have a Yeastar S100 PBX and also have Zoho CRM. I wanted to integrate the Zoho CRM/One with the PBX so that the calls made to customers outbound and inbound etc gets logged. Simple job for someone who knows how to do it.

Our main goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (Dinstar gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted ...

We are trying to build a new Asterisk server using PJSIP instead of the older SIP module and need some help to make it work. The setup is as follows: Asterisk certified/16.8-cert3 running on CentOS 8.2 with a public IP address. Phones in the office need to connect to the Asterisk server, they are on a private LAN (192.168.1.x/24) and access the Internet via a NAT router. Two types of phones: Aastra 6757i and Grandstream N300. The server also needs to send and accept calls over a SIP trunk to another Asterisk server also on a public IP (PSTN Provider). You will need to be an expert with Asterisk and PJSIP and work with us to get this working and explain what you did - you can have SSH access to the Asterisk server if required. For the right pe...

...(Included) Protection against toll fraud, and spam calls from the internet. (Included) AES 256-bit encryption. (Included) Secure Shell Version 2 SSHv2 only management. (Included) Username/Passwords encrypted. (Included) All un-secure and vulnerable protocols disabled. (Included) Brief Features Summary: It is configured for internet access, Cisco Unified Communications Manager Express phone system (IP PBX), Quality of Service for VoIP, and data/internet. Cisco 797X and 796X phones 1st button configured as the person/extension, 2nd button as a group paging/intercom, and the rest for personal speed dials. Dialing Rules Three analog extensions: 110, 111, 112 3-digit extension numbers from any phone: 1xx VoIP Extensions: 101, 102, 103, 104, 105, 106, 107, 108 Virtual extensions: 1...

...es/BMO/(124): FreePBXmodulesCertman->__construct(Object(FreePBX)) #4 /var/www/html/admin/libraries/BMO/(37): FreePBXSelf_Helper->autoLoad('Certman') #5 /var/www/html/admin/libraries/BMO/(298): FreePBXSelf_Helper->__get('Certman') #6 /var/www/html/admin/libraries/BMO/(39): FreePBXHooks->preloadBMOModules() #7 /var/lib/asterisk/bin/retrieve_conf(81): FreePBXHooks->updateBMOHooks() #8 {main}...

...(Voice, Video, Messaging) - Cloud PBX Untuk pengguna yang tidak memerlukan perangkat PBX di perusahaan. Akses layanan dapat digunakan melalui ponsel mobile apps di ponsel berbasis Android, IoS (menyusul), melalui PC/laptop Windows atau Mac, dan melalui IP phone, sehingga user experience yang dirasakan adalah Fixed-Mobile Convergence Mendukung fitur umum PBX seperti IVR (interactive voice response), call forward, parallel ringing, dsb b. Layanan SIPTraaS (SIP Trunk as a Service atau Cloud Trunking) Untuk pengguna yang memiliki perangkat PBX atau VoIP Gateway di perusahaan dan ingin menghubungkan jaringan teleponi perusahaan dengan VOXNET Dengan bergabung ke dalam VOXNET, sebuah perusahaan bisa menggunakan telepon kantor yg terhubung melalui PBX...

Hello, We need to create extensions and edit some params through an api. The parameters we need to be able to edit/create/delete are; -Extension number -Extension alias (display name) -Direct DID -Outbound DID -Inbound Direct DID -Password -Follow me settings (initial ring time, followme list, ring time, confirm calls) Our Asterisk is an Issabel distribution.

We need to develop a softphone application to work with Asterisk freepbx.

PBX is functional with 1 softphone, and 2 cordless yealink phones connected. trunks function properly. Poly com phones will not register, I have tried every possible combination of input into the web interface of the VVX400 phone and iI cannot get it to register. I just need help solving the problem here.

Hi, need basic twilio phone system set up - press 1 etc plus and outside office hours greeting. Will just do hourly.

We have a bug in our implementation of Kamailio + Asterisk + Mysql Proyect, we don't know why, but the Asterisk crashes with Memory problem.

Fix audio issue on Asterisk. Some calls have no audio.

3cx PBX system should communicate with IP Office 500 and able to configure the same extensions(or same series of extensions) should be able to install in 3cx Mobile App. So, while making the calls from 3cx- it should be from the Avaya Line.

First of all: please post real bids. I will not pay more than your bid. And please read the project before posting bids. Would not pay more than...provide the real data after accepting the project) - tu excecute some other commands. I can not even stablish connection. This code: $client = new PbxClientHi( 'sample-domain:16060', [ 'credentials' => GrpcChannelCredentials::createSsl(file_get_contents('')), ] ); returns server error 500 - PHP message: PHP Fatal error: Uncaught InvalidArgumentException: Message constructor must be an array or null. This is an open source messanger project that I am setting up for my needs. More info:

Hi We want to Integrate 3cx PBX with Perfex CRM Please find the details

We want to have sip trunking between Avaya and asterisk pbx.. If anyone have experience and stays in Jeddah let us knw

Freelancer willl provide expert advice for GOIP equipment and settings. Please let me know about your experience.

I need to install and configure Asterisk

We have a requirement to assist with the initial configuration of an Audiocodes Mediant 2600 SBC to connect a Gamma SIP trunk to an Avaya IP Office PBX and guidance on setup of dial plan routing. Must have Audiocodes skills.

Hola Felipe V., vi en un mensaje que publicaste que sabes de asterisk y grandstream. Tengo un problema y no encuentro la forma de solucionarlo. Te dejo un link con mi problema, si consideras que puedes solucionarlo me gustaria que me contactes y me digas cuanto me cobrarias. Gracias

] WARNING[21199][C-00000007]: func_base64.c:79 base64_helper: Syntax: BASE64_DECODE(<data>) - missing argument! this is what i get when i am supposed to receive a ussd asterisk 16

Custom Themes for Vicidial and Call flow changes for asterisk for Inbound Call Centre SIP based Outbound Call Centre Sip based IVR UI based Predictive Dialer Pls share the demo of your work on Vici Dial

To allow Fusion PBX to support more than one domain through text messaging

...MOBILE DEVELOPMENT: - - Custom mobile application development. - AR/VR Apps - Wearable Apps - AI-based Apps - Big data and analysis Apps - Blockchain mobile Apps * PROJECT CONSULTING: - You can book a free consultation of: - Projects - Process - Collaboration * Network Solutions: - - Lab setups - Firewall & Proxy - IVR & PBX Solutions - Corporate & Marketing Email Server - Back-up services - Network monitoring * SURVEILLANCE: - - Surveillance for Manufacturing - Surveillance for Healthcare - Surveillance for Retail - Surveillance for Education - Surveillance for Hospitality - Surveillance for Banking & Finance * POINT OF SALE: - - Rest...

I have a server with all my asterisk files that I need someone to fix up and make admin panel for me

APLICACIÓN SIP CALLING QUE REEMPLAZA LOS TELÉFONOS DE LA HABITACIÓN DEL HOTEL Y AÑADE NUEVAS FUNCIONES. El objetivo del proy...recepción, el spa y más. Al estar diseñado como una aplicación para un dispositivo móvil, el producto aseguraría que los huéspedes tengan la capacidad de realizar y recibir llamadas para servicios de hotel en cualquier lugar de las instalaciones del hotel. Ya disponemos de servidor SIP asterisk con API específicas para las funciones necesarias, tales como alta de cliente, alta de cuenta sip, activar/desactivar cliente. La cuenta SIP creada será la que se use para registrarse internamente el sistema en el servidor asterisk. Es recomendable nos envíen de...

Hi , Im looking for an expert in ISSABLE voip system to build an evaluation / Vote module (Web Pages) for Agents ... Example : after the agent finish the call with the customer he will forward the call to to evaluate process by forwarding call to pre recorded msg that will ask for evaluate the service on scale from 5 to 0 this records should be kept and displayed as graph / text and able to be exported as PDF , Excdel , the login to system must have multiple user privileged .

Dear PBX developers, you must have prior experience specifically for PBX installation into vtiger system. it is clear that if you know what you doing than this task becomes to you only 2-3 hours of task max...therefore, apply for the position only if you have prior preparation for this task and able to prepare your proof of ability with providing usable condition of your previous works, with a video or with an actual URL link of your previous CRM work with connectivity credentials for an inspection. The task should complete in 1 day in normal circumstances, however, 3 days may an optimal duration for flexibility. Vtiger 7.3 is already installed on my local computer what you going to get: I will be providing ssh connection for the Centos7 in fully working condition... I wil...

...a POC of using Asterisk for our phone support. I am an experienced software developer and an engineering manager, but I have no experience in Asterisk or PBX. This project is to run Asterisk on a new Ubuntu/20 server running on AWS, listen on SIP calls and handle a few basic operations: 1. Voice greeting, coming from mp3 or other audio file. 2. Connectivity to Google Voice Recognition, listening for next commands 3. Logging the recognized voice in text and in audio (mp3 or other audio file) 4. Responding with another greeting from mp3 files, depending on recognized voice (3 options: option A option B or unrecognized) Also, we own 2 VOIP DIDs in , which has SIP connectivity and DID POP. We would like to connect those, so that calls to these VOIP will reach th...

I am looking for a python programmer to ...documentation. Other than the example provided in the code, I'm not sure how other Classes should even be used. The assertion tests will (in part) provide an example of how each Class and Method are used. I would like assertion test for the module in You can find sample .ini files here: My primary interest is how to use append, insert, pop, and remove in the Category Class. I assume the module is capable of modifying sections and key/value pairs. Using coverage and pytest, your objective is 100% coverage, pass or fail for the module.

We currently are using asterisk version 11.22.0 We have BLF currently working Asterisk Feature Busy Lamp Field (BLF) so it is working but we have to create an BLF subscriber extension like this [BLF] exten => 1000,hint,SIP/${EXTEN} exten => 5000,hint,SIP/${EXTEN} exten => 5001,hint,SIP/${EXTEN} exten => 998899,hint,SIP/${EXTEN} exten => Prox1,hint,SIP/${EXTEN} But this is working. when the command asterisk -rx 'core show hints' run it shows hints for each extension This works but we want to create a simpler BLF BLF subscriber extension like this [BLF] exten => _XXXXX,hint,SIP/${EXTEN} But this is not working. when the command asterisk -rx 'core show hints' run it does not show hints for each extension We need woul...

Hello I need to setup Zadarma virtual pbx and integrate it with Zoho desk

I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inba...

Looking for someone to develop an IVR survey based on Asterisk pbx software, need guidance from scratch

Need someone to set up an asterix server for us; this would include connecting to the outbound SIP provider (twillo) setting up the GUI, some pilot groups, conferencing groups, automatic call recording, call transfer, sms response to incoming calls that are unanswered, a Grandstream DP750 endpoint, some soft phones, and probably a few other things on a Debian install this would also include some asterix education along the way .. I can provide SSH access for the system at a clean install point

We are using WHMCS for our Hosting and Data services billing, we are also looking to integrate the billing of our Hosted PBX solutions into WHMCS so we have one billing solution. We will export the CDR records out, and require them to be imported into WHMCS and the call rates to be applied to the clients CDR's. A custom bill will need to be produced in accordance with our requirements and compliance. If you have had experience in CDR and billing this would be an advantage.

Hello, I need to to know complete potential usage of the asterisk as well as setup asterisk Thanks

I need to configure Vtiger Asterisk Connector to do click to call (outbound and inbound calls). I have my Vtiger installed and my own Pbx

looking for some one to implement an app for push to talk PTT with asterisk as a back end .. The android app will use the functions and features of Asterisk to create push to talk functionality This message to call candidates : If you have no asterisk experience .. it will be difficult for you to deliver the work to be done . we are using a system open source system called Asterisk for SIP communications .. and we want to use the same system as a backend and build an android app as a push to talk app