Asterisk definity sip trunktrabajos
we have installed Kamailio with some asterisk server behind and we need some help to configure kamailio to be a loadbalancer and acting as SIP proxy to just pass all sip messages including register to asterisk servers behind we need this to be done asap
We have Cisco phone CP-8861 and we want to connect it with Asterisk
We need vTiger Asterisk connector expert to connect asterisk and vTiger CRM. Asterisk and FreePBX are on AWS cloud. vTiger is on another server and the edition of vTiger is opensource You must have experience in connecting asterisk and vTiger , please give us your previous work demo link . We have 25 -75$ budget. Keep this in your mind and bid please accordingly Thank you
i need to configure asterisk with webrtc
...using VoIP. You're a full stack developer. Requirements: - Backend: FastAPI / Python + Asterisk - Frontend: Svelte / Bootstrap 5 / HTML5 / CSS3 You will: 1. Develop the front-end by taking the mockups into account. 2. Develop the backend & the API routes & all mechanisms to make the front-end dynamic (except the VoIP related features). 3. Install & configure Asterisk with a Telnyx account to be able to do spoofing and 100 simultaneous calls. To test the numbers, automatic calls will have to be done. You must answer the some questions. 1. What is your expertise level in VoIP web apps ? An example ? 2. How will you interface the front-end receiving & sending the audio to the python backend and Asterisk ? 3. Could you provide an example of a Fa...
...We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or by using the Android / Windows Phone mobile versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project, we'll select the one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to WhatsApp gateway 2) whatsapp gatew...
Hi I need a dial plan for an extension in fusion pbx where I will have an extension as follows. I call the did number that points to this extension - the system tells me to dial a phone number for ide...number for identification (that is, what will be the callerid for the call - what identification will appear at the destination) - the system repeats the number I keyed and gives these options to confirm press 1 to press again press 2 - If you press 1 - please enter the phone number you want to call (destination). The system repeats the keystrokes and so on - if you press 1, the system dials using a trunk that you will have to set up (of my provider) - the call is automatically recorded. And I can receive the recording by email/ see and hear it in Fusion's interface. The budget ...
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I need someone to review my TLS & SRTP configuration of the Kazoo VoIP cluster software. I am having issues communicating between my Kazoo cluster install and my SIP trunk provider. The error message between returned on outbound calls is 488. The inbound calls don't appear in logs of my Kazoo cluster.
Recreate this bedroom set with few adjustments: Storage drawers on bed Updated carving design Mods the dressor Additional items items including (chest of drawers, trunk, wall mirror, walrobe, table and 2 chairs)
I need someone to review my TLS & SRTP configuration of the Kazoo VoIP cluster software. I am having issues communicating between my Kazoo cluster install and my SIP trunk provider. The error message between returned on outbound calls is 488. The inbound calls don't appear in logs of my Kazoo cluster.
I have the trunk configurated, and need just add a new route bentween others that already exists. This is freepbx calling from bitrix. Inbound routes are already working.
Looking for person to assist us in purchasing SIP number telephony at Task requirements Be a citizen of Germany. Have residency registration and utility bill in either of these 5 cities: Berlin, Hamburg, Munich, Cologne, Frankfurt Paying $50 for your help. If you are interested, please message for further details. Please mention & "desc" in your response so I know that you have read the message and you are not a robot response.
Dear Amrit, we changed the login configuration of the mobile application and this project is catered to support the work of this configuration. With the configuration, we will look at the app enabling SIP TLS and SRTP. We also added the domain setting so that user can change that domain setting. Thank you. John
we are required a Asterisk developer to customized the asterisk software and develop the Asterisk based pbx
We need a tecnical person who can configure FreePBX. 1. Configure SIP server to connect our SIP phones in multiple locations in WAN 2. Configure BSNL Telephones sip account to the FreePBX 3. Configure Video conference and video calls in FreePBX 4. Configure internal call transfer, call barging, recording and follow back 5. Configure all other mandatory settings. Note : We have already installed FREEPBX in our VM
Hi I need help with various settings on the subject of voip
Hi I need help with various settings on the subject of voip - required skills - diligence, flexibility, understanding, desire to progress and learn new things, consistency, availability
Develop a high-quality backend voip call mobile application on Dart. The frontend is written in flutter. Sip server asterisk install on Linux servers in AWS. We need to write the backend logic: users buy a monthly unlimited conversational tariff. Payment via api acquiring. Develop SMS to confirm mobile number. The backend code is written in Dart, read this information.
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Hi friend, I need build Opensource library with - newest PJSIP library 2.12.1 - Support full codec - Support ipv6 - sample IOS project Swift code use this lib to connect our sip platform ( will give account demo ) Support both ipv4 & ipv6
Home will be located in area code 54868 (Rice Lake, Wisconsin) SIP Panel House Design Looking for a one-level home design using Structural Insulated Panels and utilizing passive solar window design House will be built on a concrete slab and will have in-floor heat There will be no basement We would like a design that has three sections to the house. Shaped in a south-facing “U”or “V” design With an outside courtyard or gathering space in the front (south side) of the great room/open kitchen area – this area would also serve as the home’s main entrance 1. The main section has the great room and open kitchen (There needs to be room for a loft in the main section/center) 2. Master suite section – bedroom and full bath 3. 2 ...
I want to integrate FreePBX and in the future maybe other PBXs like a clean asterisk to a Billing system so my clients will be able to get their recordings, balance and be able to pay through the same platform. At the moment I have an MagnusBilling system that is not integrated yet, Maybe we need to create a new custom system or use the MagnusBilling system. I'm open to suggestions. One thing which is a must- English conversational level. I've had a bad experience because of that. At the end of this project, I want my clients to be able to gather all the information I've mentioned above by going to the link of the billing system which might or might not be the connection of the extensions. The budget of the project will be discussed as we discuss further details a...
Hello, Came across your profile during the search and I am looing for a professional who can create a sip server with APIs which will be use to make incoming and outgoing calls. Please let me know if you have time the we can discuss in details. Best Regards, Rohit
...have seen pings missing even on single host right after it's installation so I mean without even distributed switches, kernels, nics etc... configured even and still missed ping issue was there. Hosts are connected to mellanox ethernet L3 switch. And on this switch ports there are no errors reported. So It is unlikely it is mellanox switch issue. Each esxi host to mellanox connection port is a trunk port and VLANs are configured for each network such as VM network, Management network, vMotiion and vSAN networks. There is no ping issue outside VMWare network so at this point I am thinking that there is some issue within the VMWare network. And since the issue shows up in each host, it doesn't look like it is hardware related or NIC related. May be some setting or some...
I already have vicidial PLEASE READ: i need someone who can login and set up trunk for twilio and add it to my vicidial this is basic work and I need it done fast do not try to charge me a crazy amount this is very simple job I just don’t want to wait til Monday to do it please have an understanding of twilio or do your research before bidding this is for outbound call center (calls need to be recorded and showing the customers number when transferring NOT the centers) use the phrase word “I can do this” to let me know you read and understand what I need more so can do the job quickly give me an accurate price and time frame for this job
i need someone to install for me the best for 500-1000 cps , i need to put a blacklist of 1 millio number i tried with mysql it goes down any suggestion
...people of all ages. They can almost be a little cartoonish looking and will not be necessary to show many full bodies, some of them should just be faces with their hands peeking out from around the roots. Duplicates of some faces would be ok, afterall they're the same family. At the ground level, it doesn't need to be a perfectly mowed grass lawn, some roughness is probably preferable. The tree trunk does not need to be remarkable, but fairly large and old looking. The family goes back to 1400 A.D. The tree's canopy of leaves should be a sort of an abstraction of colors indicating an indefinite future that will also add more color as a background for the title to be superimposed. Overall the look should be approachable and friendly. Whether you choo...
Graphic artist required to create a graphic drawing impression of a monument to be erected in the middle of a small roundabout. The structure is depiction of tree trunk rising to FIVE branches. There is a connected pedastal on top which will support the statue of a Lady. The statue has already been made.
I have a freepbx 15 box that WAS on a DADHI channel. I need to move it to my SIP Voip provider and have another working (freebpx 13) box that is configured for my provider as an example.
I am trying to send and receive SMS/MMS messages via a FreePBX softphone when using Flowroute as my SIP trunk provider.
We need to have a Sipwise C5 CE server installed and configured. Below are the mentioned scope: 1. Installation of the Sipwise C5 CE server. 2. Configuration of the domain/reseller/sip trunk/routing etc. 3. Make a test call. 4. API walkthrough. 5. Provide us a documentation. Please only bid if you have prior experience on Sipwise.
i have 500 sip sessions in one account, Asterisk IPPBX server is alredy setup. in which i want to apply separate policies for each session.
Hi I noticed your profile and would like to offer you my project. Require custom PBX software built on Asterisk platform including GUI. It will be limited to number of users/phones and license to be uploaded/entered for upgradation of users/phones - IVRS and log - Phone Logo change - Live Monitoring and real time status - Video Conferencing - FXO/FXS/PRI/Trunk Gateway Integration & support - License based features to limit number of users/phones is a must
Hi Eremin P., I noticed your profile and would like to offer you my project. Require custom PBX software built on Asterisk platform including GUI. It will be limited to number of users/phones and license to be uploaded/entered for upgradation of users/phones - IVRS and log - Phone Logo change - Live Monitoring and real time status - Video Conferencing - FXO/FXS/PRI/Trunk Gateway Integration & support - License based features to limit number of users/phones is a must
Goddess playhouse x plush pouches presents: Pink Plush Fridays A Paint & Sip Experience 9/9/22 6pm Ticket price $45 Drinks, food, music, gift bags
...in french, which means "The Naturopathic doctor". It's about personal healthcare following natural rules - applied to nutrition, physical activities, detox, relaxation, massage, water treatments, ... We don't want the traditional caduceus (2 snakes) - but rather something that could be based on the olive tree. We have a very old olive tree in front of our building (with a short and crooked trunk) - see picture. The fonts should be fluid and harmonious (inspiring well-being). More curved, no real straight lines. Colors: olive green (light and/or dark) - we like gradients as well. background could be white (on paper documents) or a very dark greenish colour for the website, almost black. "Le Naturopathe" should be fully written with tha...
We have a HP DL360 server with vmware with Digium E1/T1 card. We wish to configure the same as our PBX for PRI line and configure all features step by step as needed. Guidance on correct setup is expected from the freelancer.
Someone must get experience in FreePBX and Asterisk. Main task is making dialplan , and solve some small issues. if someone have rich experience in this field, it will not take 3 - 5 hours. Long Term Project. As based on this result, we can work continually.
hello I have a look up numbers app now , but I wanna do some upgrade to it ,which is adding voip calling and texting feature. so must have knowledge of voip ,sip .php uniapp . and the app will be for android and apple please contact me . thanks for you reading my ad Here is a link to my existing app: safety-number
Looking for a person who can assist in buying SIP Telephone numbers at Zadarma.com. Requirements: You have to be a citizen and have residency in one of the countries from the list below Please message with "read tz" words in your message so I know that you have read the message before applying. Further instructions will be sent in private message. Paying $50 for your services. Country list: Kazakhstan Germany Poland Italy Spain France Greece Bulgaria Ireland Belgium Austria India Hungary Turkey Georgia Czech Republic Indonesia Norway Egypt Iran Iraq Pakistan Mexico Slovakia Albania
This project requires a technician that is very familiar with a Mitel Office 250 system and has experience installing wireless SIP phones on a commercial WiFi network. If you are not familiar with the Mitel systems, please DO NOT BID. We purchased a brand new Mitel Office 250 system and already had previously attained 2x Mitel 5624 v2 phones. Mitel has confirmed that everything was compatible and we also purchased the required licenses. We require you to get the two 5624 phones working on our system. I am a system Administrator so I am not having luck making them work. Everything else is currently fully functional. Here is the current status: Phones were proven to work fine when installed on a “Simple network with no VLANs” but I cannot get them to work on our curren...
I have a number of small companies Have simple IVR set up Use Speec...same. 3CX system currently resides un Vultr cloud. About 5 companies and different persons. Most use cellphones. Now.... we will pay AUD2500 plus freelancer fees.... plus require ongoing support at extra cost. Will make part payments but only when agreed modules work and tested to work. I dont have time to spoonfeed an inexperienced party... if you really know 3CX or prefer to switch us to Asterisk .... and can understand and have handled the requirements i have set up. .. this is for you.... for an absolute demonstrated example, will pay extra now. ..ie.... you show me you have this type of system working.... note.... a work around on the screened/announced calls for both Voip handset and Cellphone is com...
Need a full time asterisk and php developer in Pune with Desktop handling engineer
Need to know DID number - who is callin + Need to know IVR destination - which button pressed (1/2 or Office/Helpline)
not able to hear real ring back tone when calling out on a SIP trunk