Chào các bạn, hiện tại tôi đang rất muốn kết nối vtiger opensource 7 với tổng đài assterisk, nhưng tôi không biết làm việc này, rất mong các bạn hỗ trợ tôi kết nối giữa 2 ứng dụng này, tôi sẽ trả phí kết nối cho các bạn. Trân trọng cảm ơn!
Hi nttranbao, I'm a newbee asterisk. Now, my project are doing function monitor video call with asterisk. Do you do that function? If you can, you can send email for me ( [login to view URL]@[login to view URL]). Thank you :) Hy vọng anh biết tiếng Việt. vì tiếng anh của em không tốt lắm. Em mới nghiên cứu asterisk và hiện tại dự án của...
...as 'Before'. One SIP phone however is very special. It doesn't have a transfer button. So no call transfer can be initiated on the phone. It should be possible however to initiate a transfer via DTMF tones. The current PBX (main PBX) doesn't support tranfser via DTMF. This is the tricky part: I *think* we need to put asterisk in between the main PBX
Find at least one human cell line that is commercially available, with a homozygous deletion of the whole tp53 gene or at least of the second and third exon. Send me the name of the cell line along with the publication/data confirming the homozygous deletion. Some cell lines have conflicting data, I cannot use those. Some valuable resources: COSMIC
I am looking for an expert programmer in asterisk that can set all call rejection codes to 503 and Instantly reject all busy circuit calls without any delay by removing the played busy circuit massage, these mods needs to be done on 2 Freepbx Server running Asterisk 126.96.36.199
i got this error in asterisk and freepbx under centos "The number you dialed is not in service, please check your number and try again" im using twilio sip trunk, please i need soeone who fixed, everything i twilio is already done
I have a spreadsheet with 1796 rows of data, each line all in one cell, like this: 06/12 JIFFY LUBE #1764 ENCINITAS CA 158.20 I need someone to go through and manually add a semicolon (;) between the date and payee, and then the payee and amount. So it looks like this: 06/12; JIFFY LUBE #1764 ENCINITAS CA ;158.20 Then to use the Date > Split text
Zoho's phonebridge plugin supports up to Asterisk 1.4 [login to view URL] With modern asterisk versions (11 and up) it's not working at all do to (apparently) java problem. And the property in the phonebridge adapter seems like it is not being set properly. With Astersik 1.8 it was working partially
...project: A mobility load-balancing algorithm is proposed for small-cell networks by adapting network load status and considering load estimation. Handover parameters are adjusted based on the overloaded cells and adjacent cells. A resource block-utilization ratio is defined as a measurement of cell load and an adaptive threshold is employed to determine overloaded
Our store name is going to be named [login to view URL] and we would like a brand new logo for our store. I have attached some of photos of the general look of our website and the current logo we use. We need something fresh and innovative. The color of the logo must be the same green code as the one in the website.
I have installed my amazon ubuntu server and asterisk server . Asterisk server is working well . I am looking for somone that can install and connect SIP so that I can send sms code perfectly
...on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud
...page is to be seen or the program is to terminate. In addition, parts that have quantity on hand values that are equal to or below the reorder point should be flagged with an asterisk. Write this program as a C++ program using structures that have bound methods, functions. Write a structure, **struct card**, that will represent a card in a standard deck
I want to edit the php website, use the cell phone number register account, use the cell phone account login and get the new password, I have twilio account and use twilio to send the activation code registered account . website: [login to view URL]
A company is running Asterisk with an onboarding system. We are looking for an engineer who can support Asterisk and Kamalieo system in the cloud. Engineer is required to set up an onboarding system for clients and support them on daily basis. Here is the following job scope. 1) To set up and maintain an onboarding system that allows a user to sign
hello, we recently had some work done by a freelancer on our bespoke asterisk voicemail application. however, customers are reporting various problems: connectivity, hang-ups, can't make changes, etc, etc.. : something is definitely wrong with what was done. we need an experienced troubleshooter to take a look and make changes to a live, working system
hello i want to make a dynamic filter with asterisk follow this features - the calls come from customer pass through asterisk filter to check if it's an spam call before to send to gateway gsm if the call is a spam, then the call no go to the gateway but if the call is a real, then the call pass normaly
Hi I need someone who is diligent with Voip Asterisk. We have a Voip freepbx setup on 1.8 For some reason Iptables is blocking our Phone lines. I disable iptables service. All is working 30 min later it automaticaly reenabled iptables no more Phone. So as the freepbx is old. I want to reinstall a new server. We have 2 tenants 5 trunks and 11 phones
I tried the Proxy Server, Socks Ultimate and Server Ultimate apps on the Google play store but none of them worked. I assume its because my carrier has blocked the required incoming ports for those apps to work. Need a solution fast. Will be hiring a freelancer today. Thanks for looking
...domain and forward all registrations and invites to and from multiple asterisk servers. I think this can be done with Domain and Dispatcher module. I have written some of the config to handle registration, but when a call(INVITE) comes in it is not forwarded to the same registration server. I have a budget of £100 for this project. If your config works
I offer VOIP telephony services and development of telecommunication solutions, I need a logo and a brochure for my site currently Im using logo which is n...logo and a brochure for my site currently Im using logo which is not mine. Send sample on your propoersal based on the description of my needs This is my site [login to view URL]
We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.
Android app for cell phone monitoring: The application will allows user to monitor your children, prevent data theft and supervise the productivity of your employees. This also can be used as kids monitoring software that I can install on any android device or kids android wrist watch. It should have the below features. [login to view URL] root access should
Cell phone app
Need to configure 10 internal accounts on IP phones, connect 3 external telephone lines. Set the voice greeting and the voice menu according to the technical task. Implement a CDR report and record conversations.
I have 2 asterisk (FreePbx) servers up and running in 2 different locations connected with an iax2 trunk. I need help with the following: 1. Install and configure chan dongle for use with Huawei K3520 in one of the locations. 2. Dial plan between the 2 asterisk servers, depending on dialed number the call should be sent to correct dongle. I have a
...application working in APK format Full source code Simple manual for compiling and generating the application from source A SIP client running on another phone that connects to the server via WIFI and able to demonstrate the functionality in this project. Features - Route call from SIP to GSM - Route call from GSM to SIP (preconfigured
Fix an existing Asterisk PBX system with WebRTC. Right now calls don't come in because FreePBX/Asterisk doesn't register with DID supplier. To make it register, some changes should be made. The system consists of 3 servers, Apache, Asterisk and MySQL. You must be able to work with Teamviewer !
We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : [login to view URL] regards
Create and train italian acoustic model based on CMU sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should be con...sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should be configured for telephone (8 khz), in order to integrate it in Asterisk with mrcp server.
I wish to have a software that can triangulate location of cell phone via gps without having to install app on the other device. I will accept other methods such as sending fake sms, whataspp message to be able to obtain gps location of cell phone. ***** IMPORTANT ***** Payment will only be made once application has been tested and meets my needs
...want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows. The server side it's done and use Asterisk server with PJSIP. I would like that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features in the 1st step of
...company offering voip phone service to small businesses. I have just a few clients. On occasion, I go out of town for a week or so, and I need someone to handle the very few tech support calls I may get while out of town. Might go the whole week with none sometimes. You need to be very knowledgeable of Freepbx and asterisk in order to support my