Sip softphone trabalhos
Hi Henry T., I noticed your proposal for project https://www.freelancer.com/projects/Linux/SIP-Whatsapp-gateway/details Can you please do the same work for me? I need VoIP Whatsapp gateway module for Asterisk or Freeswitch.
Need to make Cisco504G, Linksys 942, Fanvil X7 Series and Grandstream phones compatible with FusionPBX. Set Up Provisioning to manage… Sip Registration Local Date & Time as per Client’s Location LCD Screen & Screensaver (On/Off/Timer) Idle Key List Missed Call Key List Off Hook Key List Dialing Input Key Progressing Key Connected Key Start Xfer Key Start Conf Key Conferencing Key Releasing Key Hold Key Ringing Key Shared Line Key BLF Key Programmable Keys -Intercom -paging - Call Park - Barge In For all 3 phones, extension unregisters randomly Cisco SPA504G & Linksys 942 Issues Bring Day/Date/Time from Context Attended Transfer doesn’t work Blind Transfer Works Program Intercom Button with *8 Grandstream 2600 Series Ringing Issue, call doesn&rsquo...
Looking for twilio expert who knows how to create sip end points remotely. or knows how to work with twilio programmable voice api's
Hi Swapnil thanks for configuring USA SIP Trunk I may need your help tomorrow when actual operations start.
...and advising on the solution Logging and keeping records of customer queries Analysing call logs so you can spot common trends and underlying problems Updating self-help documents so customers can try to fix problems themselves Testing and fixing faulty features You should have the following technical experience: Linux and being able to install software and Troubleshoot on Linux SIP trace and understanding of SIP protocol Experience in multiple class4 softswitch vendor Experience in SQL and or postgress...
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: h...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: I may probably also need help with G729 codec configuration
I would like to setup a sip trunk inside twilio for outbound calls only, with a different cheaper provider (zadarma for ex.). If its possible transcribe all incoming and outbound calls to portuguese (brazil) and sent to AmoCRM(OpenAPI). Also i need to setup a IVR that has a greeting welcome message and then if it rings about 8 times and nobody answer, then the ivr takes again and a recording says: All operators are busy right now, please press 1 to receive a Whatsapp Message and continue (if it's on landline, manually insert mobile phone) and send the template automatic or Press 2 to receive a call (then integromat create a task (list) into AmoCRM(openAPI) to a user to call this contacts that we have missed calls.
We have bought sip trunk with FPBX. Now we need to deploy the pbx on a cloud. I understand the configuration but need someone with experience to guide us. The main work is to configure outgoing routes, maybe incoming routes. Once that is done we want the application to communicate with the server for phone calls (this we will do ourselves). I need a dynamic and quick to respond personality who can finish it ASAP.
Hey I have a 17 user environment that I need help setting up. My preference is 3cx and I will be hosting it in google cloud. We have 17 users spread over 2 sites currently, however, we have 32 sip numbers. Each employee has a direct line, then there are 2 receptionist phones (as well a 1 wireless phone for when the receptionist has to leave her desk) and the rest are on standby for expansion. I need assistance in provisioning the desk phones as well as setting up the call routing for the PBX. As far as features are concerned I need most of the Pro features enabled, especially the switchboard feature (receptionist desk settings) I am assuming from the feature list that I need the Pro version but I will be happy if I can get away with the standard features. I have listed t...
Hi all, I am looking for a system admin to install A2Billing on my CentOS 7 server. I have tried installing it using the following guide but it did not work: I am receiving the following error when I make a call from a DID number: pbx.c:4458 __ast_pbx_run: Channel 'SIP/DEFAULT-TRUNK' sent to invalid extension but no invalid handler: context,exten,priority=a2billing-did,s,1 I have tried this installation guide and some others on 2 different servers: one with Asterisk 16 and FreePBX 15, and the other with Asterisk 13 and FreePBX 14. I'm looking for an engineer with A2Billing experience so that to work on this task of installing A2Billing. It will be for the engineer to choose the server to use from the two I have mentioned above
We would like to connect one Microsoft Teams user to a DID on an Asterisk box via SIP
I need to configure my IP phone Cisco CP 9971 to work with SIP provider
Hi developers, We are looking for people who have experiences in the integrating Zendesk with SIP Softphone. We are in the midst of development and would like a hand to help speed up the development. Please refer to the requirement doc attached.
Hello We have an urgent need of a person who can handle Issues / Queries and enhancements in Vicidialer and also able to handle Connectivity with SIP/EPBX/GSM Gateway Connectivity with VICIDialer
Need a windows application that connects to sip server and sends fax.
i am use nutshell crm ,and CounterPath bria5 softphone, have possibly integration two system? Contact Synchronization - Inbound calls from external numbers trigger a contact lookup in CRM, and contact details are added to nutshell Contacts. This way, the caller name is automatically shown in your phone display when you receive the call. Call Pop-ups – the customer record is brought up to you automatically when you receive an inbound call. Call Journals – Calls are logged as call records in the CRM. Create a new contact automatically when a call is received from an unknown number.
Hello, I want to build API [Java or php or C/C++] to call to Asterisk Server by using Android also from Windows [Browser]. The API will be one for both Android as well as Windows browser. I have purchased Asterisk SIP server users [Two number] and they gave me two extensions alongwith Server IP and password. Will shore the details after award the project What I need a API to integrate with Android as well as call from browser. The requirements: 1. Need API to Generate a call and receive a call [Both from Android and Windows Browser]. 2. Call recording 3. call Duration Log 4. Source Code after checking the Demo.
I am looking for Linphone branding, later will do is an open source SIP Phone, available on mobile and desktop environments (iOS, Android, Windows Phone 8, Linux, Windows Desktop)
...such as houses. The walls of a Panelok building are made up of structural insulated panels (SIPs). A typical SIP has a core of insulated material such as expandable polystyrene (EPS) with a wall cladding material fixed by adhesion to either side of that EPS core. Panelok manufactures its SIPs in 4 standard widths – 300mm, 600mm, 900mm and 1100mm. Each time a builder sends our factory an order for a building, we produce a 3D model of the building in Sketchup. From this we can export the construction drawings which shows all the internal and external walls of the house broken down into these standard SIP widths; with a few non-standard width SIPs. Typically, over 95% of all the SIP panels in a building will be these standard widths - with only a few &ldquo...
...necessary guide on the system installation,configuration and usage. In summary after the configuration Vtiger users must be able to make outgoing calls, receive calls, and have access to recorded calls. Note: users in crm & freepbx soft phone extensions are already made. Only linking between the 2 servers has to been done in such a way that every crm user's call is routed through the particular softphone extension which is assigned to that user. Also while implementing click-to-call small change needs to be made in the crm which is that the mobile phone field should be masked, last 4 digits of the mobile phone should not be visible to the crm user to increase data security....
I have a working Asterisk 13 phone system, but my Trunk provider (Flowroute) recently retired their POPs and forced everyone to use PJSIP vs SIP protocol. This has literally shut down my phone system. I tried to build a new system from scratch with the latest Asterisk 13 source code, moved over my dialplans to it, etc. but now I can't get any luck with authentication between both my phones and my Asterisk server and the trunk provider. I see traffic coming from there successfully when a call is made, but it is being rejected with a "404 Not Authenticated" response by my Asterisk server. If you are a seasoned Asterisk expert, specifically with experience using Flowroute, this is probably a simple support job. If you are not, please do not bother responding to this...
Build audio and video conferencing site with webrtc and sip using jitsi. Front and back end with good management system.I am looking honest qualified developer who wants to work long term to combine these 2 sites. and blogtalkradio.com. with video conferencing part of and some features from .there are very simple but i need honest reliable people i can work with. NO PREPAY OR MILESTONES. Milestones are total garbage. 99.99% of a project is 0 to me . So if you are qualified and NOT a crook. Lets talk. Your bid is the final price. I will not pay a penny more.
hi my name is kuo ,and my english is pretty basic.. i found this project. https://www.freelancer.com/projects/python/Odoo-Call-Dialler-Click-Call/details use bria5 softphone integrate odoo 13 community. contacts i have this request. I want to use it myself.
Part 1 6 hours The following setups must be implemented within a zoom session, standard working hours 8 am to 5 pm Interventions with down time 6pm to 7am Monday to Friday Installation of a 3CX telephone system appliance 1.0 hours Integrate the SIP trunks for 0.25 hours Connection of SIP phones Yealink T42S Supported 1 hour Fax numbers set up 0.5 hours and Auerswald COMfortel 1200 IP (not directly supported but possible) 3.5 hours Part 2 4 hours Integration of ISDN connections from a Fritzbox - Providing the ISDN connections as a SIP trunk in the Fritzbox 7590 Integration of the Siptrunks in the 3CX telephone system Part 3 10 hours Transfer of all phone numbers and configurations from a Commander 6000 telephone system to the 3CX telephone system - Conversion of al...
This is to call Spain customers to ask for feedback on their recent purchase and offer a discount code on repeat purchase. There will be hundreds or even thousands of customers to call. You need to have good connection to Internet and use softphone APP to make the calls. We will provide the resources. Previous customer service experience is preferred but not a must. We pay by number of calls completed.
Hello Want to develop API fot nat traversal. My client is behind nat firewall. Client (softphone) want to connect to sip server by resolving nat as well as dynamic ip address.
SIP SoftPhone for Salesforce Salesforce integration with the Mizu WebPhone
...while the call between the external number and the ext 100 continues. In some types of failure, a graceful error would be nice. The second internal extension should get the original caller id information passed through as is. I will handle the crm integration. When the script is called, i can pass the ip address of the user calling the script. The script will have run a asterisk cli command (sip show peers or similar) to retrieve the extension and or live call associated with the ip address. If there is a asterisk config file modification needed, the change must persist changes made by the freepbx gui. Part 2: I need a method to transfer a call to a wav file through the asterisk cli or other simple method. The intention is call the asterisk CLI from a simple php exec s...
professional with great experience in Asterisk, PABX, predictive dialer, softphone and wertc
Por favor, Cadastre-se ou Faça Login para ver os detalhes.
I am not strong on Python and need help (share screen and give control to VM) Environment: Ubuntu 20 LTS, Python 3.8, Asterisk v16 Note: 1. Asterisk (works OK - Inbound call, plays monkeys via a context with an external softphone) GOAL: 1. A "hello world" python to play the "monkeys" prompt on an Inbound call using Asterisk ARI REST API. 1. Using this Python module ARI 2. See "Hello world example" My basic issue/problem >>> import ari Traceback (most recent call last): File "<stdin>", line 1, in <module> File "<frozen zipimport>", line 259, in load_module File "/usr/local/lib/python3.8/dist-packages/", line 8, in <module> File
Hi there, We need a service of configuring SIP conference bridge for BigBlueButton. Our server is behind NAT, and we can't reach the result - system always says incorrect pin. Would be grateful if you can correct it.
Hello , We need Android and iOS Sip Mobile Dialer with Integrated Tunnel/Bytesaver to work in blocked country like UAE Middle East .
Dear developers looking for a developer who has already developed or have good experience on VoIP sip mobile dialer and did .we need a mobile app where people can buy DID numbers and then use them as 2nd phone and make international calls on cheap rate through that app
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: 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:
Receiving a SIP/2.0 420 Bad Extension from Kamailio on register keepalive The first register is OK. After 120 second i got a 420 Bad Extension. ******************** REGISTER sip: SIP/2.0 Via: SIP/2.0/WSS ;branch=z9hG4bKFNPKsWQBArkN7rQwDnIuoH4p1ZCSJS8q;rport From: <sip:991125943220104@>;tag=qMzPBHwxnsxK3XAQVepG To: <sip:991125943220104@> Contact: <sips:991125943220104@;rtcweb-breaker=yes;transport=wss>;expires=180 Call-ID: fcde2b26-a2cb-7809-1606-7f7183736053 CSeq: 45151 REGISTER Content-Length: 0 Max-Forwards: 70 Authorization: Digest username="991125943220104",realm="",nonce="xxxxxx",uri="sip:",response="xxxxxx",algorithm=MD5 Organization: XXX User-Agent: XXX webRTC UA 1.0. SIP...
Build audio and video conferencing site with webrtc and sip using jitsi. Front and back end with good management system.I am looking honest qualified developer who wants to work long term to combine these 2 sites. and blogtalkradio.com. with video conferencing part of and some features from .there are very simple but i need honest reliable people i can work with. NO PREPAY OR MILESTONES. Milestones are total garbage. 99.99% of a project is 0 to me . So if you are qualified and NOT a crook. Lets talk. Your bid is the final price. I will not pay a penny more.
...calls through SIP and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Summary The goal of this project is to develop an Android application that can send calls through SIP and forward them to the GSM network AND receive calls from GSM and forward to SIP. The application should then forward the audio and convert from VoIP to GSM and vice versa General Deliveries The 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 fr...
This is is to install and configure FreePBX on a VM. Below is the list of objectives. - Install FreePBX on a VM - Configure network setting and activate. - Connect to Yeastar S100 IP PBX for outbound calls. - Help to configure an extension. - Help Setup an IP Softphone (any free softphone) and make an outbound call. If you have any questions please ask me.
...necessary guide on the system installation,configuration and usage. In summary after the configuration Vtiger users must be able to make outgoing calls, receive calls, and have access to recorded calls. Note: users in crm & freepbx soft phone extensions are already made. Only linking between the 2 servers has to been done in such a way that every crm user's call is routed through the particular softphone extension which is assigned to that user. Also while implementing click-to-call small change needs to be made in the crm which is that the mobile phone field should be masked, last 4 digits of the mobile phone should not be visible to the crm user to increase data security....
OWN HOSTED CLOUD SERVER WITH IP-IBX SYS WITH CUSTOM FEATURE , SET CALL ID , CALL OUT , CREATE USERS , KNOWLEDGE OF SIP TRUNKING
Looking for an experienced twilio person to help with configuring a new sip server on twilio to work with Linphone softphone. Immediate work
Need to build a server client application for SIP TLS and RTP MUX between 2 systems, whereby 1 box is behind NAT. TLS transport layer security will be used to encrypt SIP signaling and RTP MUX – RTP content for concurrent calls between 2 peers are transmitted in a single UDP packet.
Telephony mobile app for calling Card customers with a superior experience using iOS/Android. Also make use of APIs to communicate with a web, telephony and other 3rd party platforms. Unique and critical features it will possess include an embedded softphone WebRTC(prefered) or SIP based and mobile dialer.
i have my sip server on france now i want to register sip from a blockage country like UAE UAE have all voip blocked i dont want to use a separate VPN i saw lots of dialer there have proxy server and its work over dialer proxy so i need you to make something like this we will use a middle server like a proxy server like we try to register on our proxy server and its FWD to all my SIP server or we can make sip proxy or anything as you like i just want to register sip and make call from blockage country if you can make it please bid
Responsibilities Handling Linux Server Install Switches, PBX on Servers Maintain VoIP Billing Providing Support Server Error Handling Write Custom Codes for Development Requirements Minimum 1 Year experience in VoIP Good Knowledge of Freeswitch and Asterisk - working on astpp billing, fusionpbx etc. Good knowledge of mobile dialers an advantage too Well knowledge of SIP Servers Good in development of AGI,FastAGI, Dialplan, FS Scripts Knowledge of JavaScripts and Lua. Working Knowledge of LAMP Full time online position Educational Qualification - MCA, B-Tech,
hello I will offer remote desktop to configure a freepbx asterisk install. you have to add a second ntw card , do static routing and integrate sip trunk with my telephony provider
Hello, I want a developer who can create a SIP investment calculator for me. Whether it's a plugin or html page that can be embedded to Wordpress. The plugin/page will just have to calculate that If I invest "X" amount of money for "Y" amount of year and expect "Z%" return then how much will the person have that time. If you need some samples, please PM me. If you are up for the job, please PM me with your price. Regards
