We need you to install vtiger and configure it, install freepbx an configure it and implement a phone call option for different employees with different extentions to make calls directly from vtiger. If somebody calls the extention of the employee, vtiger must automatically show the contact-info of the caller (go to the respective client in the CRM)
...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 to connect to a translation service, internal or external, and route voice traffic through the
We are looking for the help of a ...points that we would like to express in our banners: - We have the best products for a 70V Paging solutions - Our amplifiers can directly connect to the Internet and offer SIP paging solution without an adapter module - Our amplifiers can all be managed remotely via a WEB Interface - We distribute 70V speakers
I need an Android app. I would like it designed and built. I need to turn an Android cellphone as SIP to GSM Bridge of Gateway. Traffic from SIP must be translated to GSM channel. You are free to choose how to implement this feature. All that is required is that in the end the Android phone will be acting as a SIP to GSM channel/trunk/GW an...
El proyecto incluye implementacion y soporte a conmutadores telefonicos Panasonic, Cisco, Avaya en protocolos R2, y SIP, y cableados de nodos de voz y datos
Hey, you did setup a long time ago vtiger with freepbx integration (direct call from vtiger wa...doesnt work anymore :-) However, I need the same work done again. Setup of vtiger on a website and the full freepbx integration, that every user can make calls with his own sip account. Can you do the setup again with the newest versions of both programs?
I need a SIP Client like as eyebeam and zoiper. My Reqs : I need to define 1. sip server ip, username and password 2. Auto Answer option (If this option enabled calls automaticly answered without press to any answer key) 3. Softphone can record conversation and agents screen desktop when call is active. (File contains datetime_callerid_agentusername)
Build a Reporting System , based on Vicidial database architeture, ...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
I have code for registering on SIP account already. But it is not working. If you are developer have experience for SIP calling on Android, please help me. I can send my code to selected freelancer. I don't want team, need only individual. Thank you.
Problem is that the dependencies don't install using sudo apt-get in the live cd version of ubuntu 17.04 Actually terminal displays package not found for sip and for pyqt-4. I need you to write in your bid how to achieve before 24 hours.
...record. The closest location geographically (MS ping) the Asterisk server provide quality service to the customer. Other routines use Trunk routes, in case of failure. And we want to use the cheapest route. iphone + android + mac + Windows voip we can think later of their software. But we can evaluate the proposals. The colors of the site design website
I need android application for door entry system. It should be built using SIP and VoIP. Design(include XML) was completed already. Don't bid without sip and voip knowledge, please. I will reject that man immediately. While chatting, let me provide design.
A STUN/TURN server has been tested to work on Android apps such as Zoiper and SessionTalk (using accounts from a specific SIP server). However, our app fails to use the STUN/TURN server correctly (with the same SIP server) and produces the following error in the STUN/TURN server logs: "error 437: Mismatched allocation: wrong transaction ID" The
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: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from (callid: email@example.com) - No matching endpoint
Sonstiges oder nicht sicher Sonstiges oder nicht sicher Quiero hacer un proyecto de un videoportero IP En Raspberry y la llamada debe ser a un android o iPhone vía SIP video allí. Lo quiero construir para fraccionamiento. En un Touchscreen seleccionar el departamento/casa/persona y Raspberry lo hace el vídeo allí al celular/ tablet.