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    8,457 sip software open source java jobber funnet, priser i USD

    i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

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    VOIP, SIP Expert 6 dager left
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    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

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    Connect IP PBX to Twilio WebRTC 5 dager left
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    I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

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    SIP Proxy Server install 5 dager left
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    i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

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    you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

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    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    Secure Issabel PBX (asterisk) 3 dager left
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    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

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    Trophy icon Création logo Startup 3 dager left

    ...companies. It's a startup centred on cloud technologies. (Voip, Sip) The name is " Caraïbes VOIP " The main color must be blue like the sky. Je démarre une startup dans la téléphonie sur IP à destinations des entreprises. Le projet est axé sur les nouvelles technologies CLOUD. (Voip, Sip) Le nom du projet est "Ca...

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    ...com/click-to-call/ etc You don’t even need a fax machine…,Receive Fax from customer as PDF in your Email. The system will support setting up/administrate the call center SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability to activate/deactivate service

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    Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable

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    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

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    I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

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    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

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    I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.

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    ...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me

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    I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.

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    WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network

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    VOIP ASSISTANCE Avsluttet left

    VOIP ASSISTANCE SIP DEVELOPPEMENT

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    Trophy icon JT&T Brand 21 dager left

    I’m looking for a logo for business cards and letterhead I’m looking for some flyers so that I can advertise VoIP and SIP services the Name of the company is JT&T Communications we offer best in class SIP and VoIP. And phone system analysis design and implementation

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    Garantert
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    ...vicidial server files ( my dialplans, modules , database ,IVR , [logg inn for å se URL] , custom code on the server, user details, ALL CONFIGURATIONS. etc ) to new & better hardware /server Xeon processor 3.2 GHz with latest Asterisk & Vicidial software installed 2) Sugar CRM Community Edition/Open source CRM to be installed and configured for flexibility to...

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    1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.

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    ...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [logg inn for å se URL] secret GZD397XU Host [logg inn for å se URL] port 5060 Register String 390415541985@[logg inn for å se URL]:GZD397XU:390415541985@[logg inn for å se URL] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [log...

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    ...for us. Do you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications. Zoiper softphone is a standalone client-side software VOIP phone application and

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    ...routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new routines quickly and efficiently. We have

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    Lightweight SIP Client with Dynamic CRM which would display the customer information. CRM Field and information would display dynamically and action should collect and send to server Database. Required Features: SIP Call with transfer/conference/hold/Mute facility SIP messenger. Dynamic CRM-Dynamic field header and value display for each call according

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    I need some one who is expert in cyber security & hacking and specially VOIP, Sip, you should be expert in this field.

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    I need professional help with setting up duplicate ...• 18 Avaya Phones – Model 1416 • 23-24 extensions Needed • 4 Trunk Cards That I see • 5 Analog Phone Numbers Following Licenses currently on system: • SIP Trunk Channels • SIP Trunk Channels • SIP Trunk Channels • Essential Edition • Essential Edition Additional Voicemail ...

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    as the title indicates, need sip installed on cisco 7960 to work with astersik. total 10 phones

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    Hi. We need to configure asterisk on our ubuntu machine. We need someone high skilled to do the job. We will also need to make it visible over internet with public IP. Also we need to configure some IP phones on it. After all, we need a manual with step by step about how to configure it.

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    I have lan calls working fine, but require someone with Cisco/Asterisk SIP knowledge to help resolve this issue with NAT, no audio not working.

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    I require a voicemail drop software platform. The application will feature voicemail detection function where an outbound call is placed from a csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as callcentric

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    Predictive Dialer Avsluttet left

    we need a Predictive Dialler with freepbx to link to Qs and have a wall-board with reports. reports need to link to recordings. this all has to work on freepbx and sip and softphones. The PBX will make the call and the agent when signed into the Q will receive a call. the agent phone rings first then the PBX makes a call according to a predefined list

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    Hi I need an application that can receive a call through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset and set it as default

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    ...to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work for both and each by separate registration. 4- send calls from sip/iax2 to gsm and receive calls from gsm and send it to sip/iax2 5-

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    ...and want to have a branded Softphone for Android and IOS. Registration and provisioning of softphones must happen with SMS verification. For the softphone we want to use the source code from VoipGrid as they support push notifications for incoming calls, have IOS and Android versions and also an existing auto-provisioning with email. So the code changes

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    ...1main server other all asterisk server work like VM host1 IP [logg inn for å se URL] SIP 5060 DB NAME datastore1 host2 IP [logg inn for å se URL] SIP 5060 DB NAME datastore2 host3 IP [logg inn for å se URL] SIP 5060 DB NAME datastore3 host4 IP [logg inn for å se URL] SIP 5070 DB NAME datastore4 host5 IP [logg inn for å se URL] SIP...

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    Hello All, We need to integrate Vonage and Callcentric SIP with our Bitrix24 installation for calling to USA. This will be used with a 10 PC user system for a call center. Other requirements with same: - SIP set up in Bitrix24 - Setting user right for 10 users with extendable option to more users. - Setting up rules for call forwarding, call picking

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    We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Proficient in Lantek program (lazer tube cutting) is a must. We have ongoing work for the right programmer on a long term basis.

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    ...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Artificial Intelligence: It involves Studying Human thought processes representing thought process on machines. Knowledge encompasses the implicit and explicit restriction

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    ...routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new routines quickly and efficiently. We have

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    URL Free to Call System from the website, to use my sip trunk for clients to call me free.

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    ...DTMF button on screen so user can press options from IVR and user use inbuilt microphone then connect call to SIP address on Incredible PBX OR When number clicked on "Free Call" button, viewer types in his/her phone number and submits to SIP address which setup on Incredible PBX Example [logg inn for å se URL] The right candidate will be also contracted

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    ccnp voice Avsluttet left

    ccnp voice training and configuration for voice gateway mgcl sip and h323

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    I need an app for Android and IOS that basically takes and makes sip calls. It can be an open source phone labeled with the name that I need.

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    ...GrandStream (con acceso al personal de soporte directo del fabricante). -Configuraciones de troncales SIP dinamico y estatico. -Facilidades de troncales SIP con las prestadoras de República Dominicana. -Configuraciones de IVR. -Cola de llamadas. -Extensiones SIP. -Configuraciones de PBX redundante Asterisk -Configuraciones de teléfonos VoIP....

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    centralita Avsluttet left

    Tengo un gateway goip gsm de 8 sim, 4 voipeadores tplink y un servidor en asterisk pero necesito crear la troncal sip y enlazarlos con los voipeadores para realizar llamadas salientes por el gateway gsm

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    Need a SIP endpoint built in Twilio to point traffic to my freepbx installation. Preferable if Twilio can only send 10 digits.

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    Hi Olexander M., I want to install a new asterisk server. We will use it as a SIP proxy. We will have a blacklist B numbers in db (or anywhere), we will sync this data every minute.
And Asterisk will check if the incoming all is in blacklist, and will route the call after. The flow would be like: call comes from A server, to Asterisk server with 3333

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    I need some help to fix up some call problem with an Asterisk server - All internal are behind NAT (with Miktotik router) - Asterisk server are on a VPS on cloud

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    We are facing some challenges regarding our "On Guard" phone services - we need a solution which can provide 24/7 Phone support, Insights and a really good customer / user experience. We have a current very old custom build solution based on a Asterisk PBX with ISDN and want to move it to the cloud and provide new features for our team. Its very important that you have indept knowledge...

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