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    14,246 sip configuration symbian tugasan ditemui, harga dalam USD
    Elastix 3.0 MT Tamat left

    We would like you debug and fix our sip trunking /Ext to Ext calling setup to work with our providers so that we may use Elastix MT 3.0 to the fullest with multiple organizations and users per organization, Elastix MT.3.0 has been installed already just need to fix some bugs. We are currently able to have our handsets receive registration, and our

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    My Name is Fayssal Daou...follows; 1- Design the telephony integration optimum system 2- Generate a block diagram and BOQ of required software and hardware (PBX, Virtual PBX, Handsets, Server...etc) 3- Configuration of the system 4- Handover Documentation including (Operation and Maintenance) Note: we will execute the procurement and the installation

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    Unfortunately my project has been delayed a month and i am ON...serious freelancers, who can complete my project within 3 days. Tasks pending: 1. Payment gateway setup 2. Shipping plugin install([login to view URL]) and configuration 3. Pickup plugin install/configuration 4. Minor translation in danish language 5. Bugs testing of complete template/website/

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    We need an expert who is having experience in Customization and Configuration in Business Central. Only Bid if you have already implemented. This is long term opportunity. Please start your BID with "DBCCC", Thanks

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    ...(administrative network). There is a firewall that prevents connections from 4.0/24 to 5.0/24 We have VOIP SIP Server running at [login to view URL] Currently the server runs well but we will like to make possible users on 4.0/24 to access the VOIP SIP Server If you have long experience working with this type of routers and you know how to make the required

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    KAMAILIO SIP + MEDIA PROXY WITH SIP CAPTURE INTEGRATION INSTALL INSTRUCTIONS + CONFIG FILES + TECHNICAL SUPPORT We need a KAMAILIO CONFIGURATION accepting calls from one NIC with public IP address and redirect passing the call to 3 different SIP providers connect to 3 different NIC’s with “local” IP’s (load balance - round robin). Media must b...

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    ...informações de cartão de crédito. Analisaremos como será feito durante o desenvolvimento; Opcional (projetado somente para atualizações futuras): Ligar para o profissional (Chamada SIP, somente na segunda versão) *Funcionalidades – Aplicativo Profissional* Login; Logout; Checkin; Checkout; Envio de fotos de atendimento com...

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    English speaking ERP Consultant to integrate the Odoo in the customer IT.

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    I am looking for a Asterisk developer to quickly update existing Asterisk code for dialer, inbound/ outbound/ press 1, and other advanced features. Need to configu...Polls AMD (Answering Machine Detection) DNC ( Do Not Call) Support Inbound Campaign Custom Caller ID Multiple Trunk Support Realtime Campaign Management Multiple Technology (SIP) RESTAPIs

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    Need help to translate several fields and to set up carriers/shipping methods to display correctly.

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    Plesk server need to correct some configuration errors When we add a new domain name we get the follow errors Error: phpinimng failed: configuration test '/usr/sbin/php-fpm --test' failed with code 78, and message: [21-Aug-2018 22:54:15] ERROR: [/etc/php-fpm.d/[login to view URL]] value is NULL for a ZEND_INI_PARSER_ENTRY [21-Aug-2018 22:54:15] ERROR: Unable

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    I need a SIP cordova plugin to use with Ionic 3. Cordova plugin is already exist. [login to view URL] All you need is the create ionic plugin wrapper for cordova-plugin-sip.

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    13 bida

    I need a little bit of help to design my simple VPN infrastructure around Mikrotik routeros (eg. IP plan and Mikrotik configuration).

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    13 bida

    ...are freeswitch and fusionpbx FusionPBX CDR billing requirements Problem statement Currently i have a single FusionPBX configured in a multitenant environment with different SIP trunks from multiple providers configured per tenant. Given that there are multiple providers, the upstream CDR reports are not suitable for processing the clients billing for

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    rebuild asterisk server using backup files sip and dialplan database AGI (python) restore all default functionalitys

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    19 bida

    i need to install correctly ssl on prestashop - it works only in backoffice, in front it shows an error, i tried to change ssl settings in prestashop backoffice but it wont work

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    ...migrated to a Cisco 3945 router and it does not work fully. I have: Cisco 3945 with Voice license 1 x 4G Samsung phone with Zoipher that registers to my SIP server over the WAN 2 x LAN connected SIP clients 1 x FreeSwitch Server on Windows You're free to change the setup, but this was working using FreeSwitch fine on my lan until i changed things over

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    I would like this software to be developed for Windows using C or C++. MOSQUITTO MQTT broker configuration with and without websoket, TLS and proxy settings. Deployment of the SW on the website provided. Testing with the embedded device. details? Detailed Requirements are: I have to deploy the MOSQUITTO MQTT broker on my website where it will run 24x7

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    We have twilio and need someone to seup the Elastic SIP Trunking and make the phones dial out and ring... via Ciscso SPA540G and obihai

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    Hi guys, I have an android app which is already completed. It has few bugs on few phones. Can you resolve them? Are you aware of SIP protocol?

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    i need someone experienced about issabel (formerly elastix) to set up an maintain our vps based sip switches. serving to multiple clients using multiple sip operators,trunk or sip user based.

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    I need an Avaya IP Office expert who is familiar with SIP Trunk configuration with fax server products. Currently we have the licensing installed for 4 SIP Trunk Connections, but the Avaya IP Office is responding with a SIP "Temporary Error".

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    I am looking for someone who can show me how to set up Mailwizz so I can start sending emails

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    ...should have a universal form of support for mjpeg H264 formats.h264+. H265+ support IPv4 / IPv6, TCP, UDP, RTP, RTSP, RTCP, HTTP, HTTPS, DNS, DDNS, DHCP, FTP, NTP, SMTP, UPnP, SIP, SNMP, PPPoE, VLAN, 802.1 x, QoS, ONVIF Profile S & G, and to integrate via the SDK P2P, video Analytics modules [login to view URL]

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    We are currently running a Cisco 2851 with Cisco CME 8.5. In order to manage better voice mailboxes, international lines for call-conferencing we want to upgrade the system to CUCM on a Virtual Machine and integrate a software like MARS meetme Password to manage call conferencing remotely, auto-attendant music etc etc.. The current CME Config is fully available and there will be available an emp...

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    I have third party Linux Server, SIP/VOIP, when an IP phones registered with server and are assigned with extension number like 501,502 etc, now these phones make communication with each other by dialing their number. What I want to do is, remove the IP phone and use Raspberry module, and it should work as phone, and connecting push button switch on

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    ...of work but getting frustrated with what looks like sip/nat issue and that I couldn't find a way to be able to call between local extensions. It's mainly for personal use, I wanted a simple PBX that could handle a small group of sip devices that could call each other and that could use an external SIP gateway. The aim was to have better sound quality

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    3 bida

    ...server and then have all SIP traffic route over the VPN to allow 5060 traffic out of Dubai which blocks SIP on 5060. Install IPSEC on the Ubuntu server (currently in production hosting web app) Configure VPN Ensure only SIP traffic flows over VPN lock down the entire solution so the SIP traffic can only go to one IP address our SIP service This is a remote

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    Trophy icon Sip and paint Tamat left

    Paint and Sip for a Purpose. It's on September 13 2018 from 5 to 8 at the Palace Bar and Grill located 234 East Main Street in Waterbury. Tickets cost $25 includes all supplies and glass of sangria. Any questions contact Denise at 203-598 1781 dmartinez@[login to view URL] Benefits NOI annual toy drive.....accepting donations

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    Dijamin
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    3 penyertaan

    Script installation and site configuration Domain connection and ssl activation Email and smtp settings

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    Hello, as the title i need someone who can connect mailchimp module with mailchimp account on prestashop website. actually i did all the things but i always see this message List Creation failed. Please check that Store Address exists. A physical address is required to create a list. More here: [login to view URL]

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    Make calculator for website SIP calculator, retirement calculator ,loan calculator

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    ...already installed A2Billing asterisk server and set up a Sip Trunk to my VoIP provider. My internal free PBX extensions are working, however, I need help in configuring the A2Billing servers. I have already set up the following: 1. Free PBX & Extension 2. Free PBX routing and dial plan 3. Sip trunk to VoIP Provider 4. Outbound dialing rules 5

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    ...already installed A2Billing asterisk server and set up a Sip Trunk to my VoIP provider. My internal free PBX extensions are working, however, I need help in configuring the A2Billing servers. I have already set up the following: 1. Free PBX & Extension 2. Free PBX routing and dial plan 3. Sip trunk to VoIP Provider 4. Outbound dialing rules 5

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    I need 4 assemblies to be created in Profile Builder version 3: They are: SIP wall (including Internal and External Panels), Concrete Waffle Slab Floor, Mid-Floor and SIP roof. Each Assembly will be treated as a Milestone (the value will depend on difficulty). Outline of each assembly will be provided, but there is a lot of detailed information (both

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    ...purchase. Project Pitch We would need an experienced UX/UI Designer to create an appealing visual mockup for our upcoming new digital tool: the Social Impact Profile (SIP). The SIP will be a user database and part of our existing website ([login to view URL]). With this tool we want to offer our customers the possibility to create their own user account

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    NaughtyTales Tamat left

    I have written erotic stories for years. Some in which I have published and others in which I have written for other ...erotic stories for years. Some in which I have published and others in which I have written for other people. I love when I get in the groove and just let my mind wander while I sip my wine and free my mind... I'm sure you will too!

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    Implement and configure the open source core banking system Mifos X.

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    Hello, We need to develop a SIP to Viber/Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/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 Viber/whatsapp

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    4 bida

    Experience person who have worked on Betfair api before can only ask for work .

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    ...[login to view URL] in remote on my servers. a lot of work. good employees can work with me for years. job: Step One. find correct configuration for the script on [login to view URL] , This should take few hours. Step 2 get info automatic from emulator (multi devices) can take 2 days of tries. STEP 3 get api commands

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    ...potential customers and setup sales appointments with upper management for website design and development. We will provide the call list of companies in our local area as well as a SIP line through Telzio to make the calls. The individual who wins this job must have good internet, at least 25/3Mbps to be able to make internet phone calls. Must possess a professional

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    I have to config...theme ... Simple Product Configurable Product Grouped Product Virtual Product Bundle Product Downloadable Product I tried to install this configuration (url indicated below), but when I go to insert the new product, it goes wrong ... thanks [login to view URL]

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    Hi. I am IT Engineer. I am working as a programmer. I have tons of application names (approximately 1200) for Symbian and I need detailed information about every of them. I wrote data collector app in C# but for me it is not enough. It uses database of one popular site and not suitable for my work. It can only find details for 30% of apps and most

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    I have a Linux based SIP VOIP Server, by registering any SIP IP phone with this server and giving them unique number say 500,501,502.. and they works as VOIP calling system by dialing the telephone number. Now what I want to design is, instead of IP phone use Raspberry Pi as a phone and when registered with server it should work as a Telephone. for

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    We need to develop a SIP to Viber/Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/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 Viber/whatsapp executables

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    Perjanjian Kerahsiaan
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    1 bida

    Looking for experienced Android developer who can instantly resolve some bugs in my App and can also configure Crashlytics at the earliest. Have to work via Team Viewer. Those who are interested and can work via Team Viewer, please submit your bid.

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    Aplikacja VoIP Tamat left

    ... Możliwość zestawienia kilku kont SIP od danego dostawcy. Możliwość wł./wył prezentacji numeru. Blokada połączeń przychodzących/wychodzących. Możliwość ustawienia prezentacji innym numerem. Możliwość rozpoznawania wybieranego numeru np.: krajowego lub międzynarodowego i nawiązanie połączenia dedykowanym kontem SIP. Możliwość zestawienia telekonferencji

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    Google Cloud Engine and cpanel needs installing and configuration with other php extensions and mail, security etc to make i ready for the sites to be installed using cpanel. Its a 2 hours job at most!

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    Server OS: Centos 7 Software: Asterisk 13 + FreePBX + a2billinng Network: Dynamic IP beh...OS: Centos 7 Software: Asterisk 13 + FreePBX + a2billinng Network: Dynamic IP behind NAT Problem: - Call receiver no ring - No audio as it should to tell balance, dial 9 for sip or iax Because this desktop behind NAT, i'll provide Teamviewer or Anydesk access

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