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    14,530 xmpp asterisk travaux trouvés au tarif de EUR

    Nous travaillons avec FreePBX (Asterisk basé sur FreePBX) depuis des années et souhaitons migrer/tester XiVO Community (basé sur Linux Debian). Dans un souci d'efficacité et de gain de temps sur la R&D (recherche, développement, déploiements, tests), nous cherchons un prestataire freelance qui maitrise bien Xivo. Nos besoins sont assez simples : - Configuration d'une carte OpenVox (4 lignes) - Configuration des lignes - Création d'utilisateurs (Client SIP) - Réutilisation d'un script (IVR) - Limitation d'appels sortants (via une Regex) (....) Nous pouvons fournir un accès distant VPN et SSH sur une machine virtuelle (ou la carte physique OpenVox est associé à la machine vi...

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    predictive dialer avec asterisk S'est terminé left

    je recherche à acheter, a louer ou à faire créer un predictive dialer pour les callcenter

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    ...exceptionnelle de rejoindre une société belge de eCommerce et une équipe jeune et enthousiaste qui connaît une croissance très rapide. Profil recherché : - Sérieux : intégrité et stabilité. Références obligatoires - Attitude professionnelle, attitude le client est roi - Autonome (capacité de propre initiative) - désireux d'apprendre - bonne connaissance du web - Connaissance technique VoIP / Asterisk / SMS / Réseaux et serveurs - Langues : français et anglais écrits. Très bon niveau exigé Description du travail via Chat: - conseil client (soit le client initie le chat, soit l'agent invite le visiteur sur le site) : connaissa...

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    Support francophone pour site VoIP S'est terminé left

    ...exceptionnelle de rejoindre une société belge de eCommerce et une équipe jeune et enthousiaste qui connaît une croissance très rapide. Profil recherché : - Sérieux : intégrité et stabilité. Références obligatoires - Attitude professionnelle, attitude le client est roi - Autonome (capacité de propre initiative) - désireux d'apprendre - bonne connaissance du web - Connaissance technique VoIP / Asterisk / SMS / Réseaux et serveurs - Langues : français et anglais écrits. Très bon niveau exigé Description du travail via Chat: - conseil client (soit le client initie le chat, soit l'agent invite le visiteur sur le site) : connaissa...

    €181 (Avg Bid)
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    Configuration d'un serveur KAMAILIO S'est terminé left

    ...numéro de téléphone SIP fourni par OVH 1 serveur dédié sur lequel est installé KAMAILIO. Mon téléphone IP était directement connecté à ma ligne SIP chez OVH (Je pense que ma ligne SIP chez OVH est installée sur un serveur ASTERISK chez OVH) Mais depuis peu, mon FAI bloque les enregistrements et appels SIP. Mon projet est de configurer un serveur intermédiaire KAMAILIO pour l'utiliser en tant que PROXY SIP pour enregistrer mon téléphone IP sur KAMAILIO sur un port différent du 5060 et que ce dernier relais les appels entrants et sortants entre le serveur ASTERISK d'OVH (sur lequel est installer ma ligne SIP) et mon téléphone IP. ...

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    La société LWS, spécialisée dans l'hébergement de sites web depuis 1999, recherche une personne expérimentée dans le cadre d'une mission ponctuelle. La mission consiste à - installer un trunk SIP sur un serveur Asterisk - installer minimum 5 comptes SIP pour appels sortants avec enregistrement de chaque appel - installer un panel de gestion (XIVO ou autre)

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    Codez un logiciel S'est terminé left

    Bonjour, Je chercher un expert Asterisk et le protocole SS7 pour mettre en place un script qui permet de terminer des appelles venant d'un serveur asterisk en utilisant un API SS7 : Trafic Voip -----> Asterisk ---->>API SS7 ----> Destination Cordialement,

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    L'objectif est de connecter Vtiger CRM cloud edition avec un AsteriskNow. Le travail est en partie réalisé mais certains détails ne fonctionne pas. Nous recherchons quelqu'un pour peaufiner l'installation. Vtiger et Asterisk se trouve dans le même réseau et l'installation reste basique.

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    Nous avons un serveur asterisk configuré avec quelque sip trunk ovh. Nous avons des problemes dans la presentation des alias dans les appelles sortant. Nous avons besoins de quelqu'un qui connait bien se fournisseur Merci

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    Excellente connaissance de la gestion des ports USB, de UDEV, de asterisk et de sous linux centos et ubuntu

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    à la recherche d'un spécialiste VoIP pour l'installation d'un serveur asterisk, a2billing et fail2ban insertion des des rate plan...

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    Installation de opensips sur un serveur et asterisk/a2billing sur 2 autres serveurs

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    Local
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    Concevoir une page web (disponible dans l'intranet de la société) pour permettre de saisir un numéro de téléphone. Ce numéro doit être utilisé par asterisk pour activer une déviation de numéro. Actuellement le numéro en question est codé en fixe dans le fichier de configuration

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    XMPP Protocol Consultation S'est terminé left

    I Want You To Create A XMPP Protocol Chart

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    Je souhaite que le site soit mieux référencé naturellement pour mettre en avant nos compétences en développement "Asterisk", "VoIP" et "CRM"...et autres à définir.

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    Mise en place d'un serveur Asterisk S'est terminé left

    LWS, société spécialisée dans l'hébergement web, recherche un développeur apte à monter un serveur asterisk complet. Une fiche explicative complète vous sera fournie. Ce projet conviendra à un développeur confirmé, ayant déjà monté un serveur Asterisk évolué : menu, logs, file d'attente etc INDISPENSABLE : Langue française (toutes les explications vous seront données en français)

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    Installer un serveur Asterisk S'est terminé left

    LWS, société spécialisée dans l'hébergement web, recherche un développeur apte à monter un serveur asterisk complet. Une fiche explicative complète vous sera fournie. Ce projet conviendra à un développeur confirmé, ayant déjà monté un serveur Asterisk évolué : menu, logs, file d'attente etc INDISPENSABLE : Langue française (toutes les explications vous seront données en français)

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    installation ipbx asterisk S'est terminé left

    Je voudrais installer un ipbx asterisk dans mon entreprise qui comporte 3 sites. Le projet comporte l'installation sur paris et un transfert de compétences pour pouvoir se débrouiller avec les pbs maintenance globale serait affectée à celui qui remporte le projet Cordialement

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    Fixing problems asterisk,vicidial,elastix voip Fixing problems asterisk,vicidial,elastix voip Fixing problems asterisk,vicidial,elastix voip Fixing problems asterisk,vicidial,elastix voip

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    he desarrollado una ivr en issabel. tengo todo el sistema de marcado gestionado con php, el problema es que cuando se ejecuta el bash para para marcar las llamadas salen pero no conectan a la ivr, nesecito soporte en este aspecto, estoy buscando especialistas en asterisk y issabel que hablen español, adjunte la consola de click y el bash que desarrolle

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    There are 3 entries in total. Each duration is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Hindi, you must als...the following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Hindi, then it is necessary to indicate “Hindi (dialect name)”; c) if in addition to Romanian there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Hindi (dialect), japanese*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    SIP with any RTP 4 jours left
    VERIFIÉ

    i have asterisk sip server and its working fine with any RTP i set on my system interface. but in same VM if i copy and make a new server when i try to set any new RTP its not working i dont know whats the issue audio is gone. so i need you to any how create a system where i can set any rtp as i want. like or or anything as i want. (we dont use our public IP as RTP we use any IP on our RTP like a eth0:1 interface ip is:1.1.1.1 so i will use this as a RTP) you can use any sip server or anything as you want. i just want to use RTP thats it. you can to setup this your local system or if you want i can give you server dont ask me any payment before test. if you can show me its working and audio is fine you will get payment with bonus.

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    Hello. I need an personalized app, a CHAT app , that should use jabber / xmpp protocol : we use Openfire server ( ) Thank you

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    Hello, we have a project need deploy Asterisk 1.8 from source code with support ODBC for user managment MySQL, and CDR record storing in MySQL, need support call recording in MP3 format and storing in with CDR details also need support video calls. * We don't allow in this project used software like issabel. * We need little bit documentate configuration of server in small text file with procedure of deployment * We need test all with LinPhone SIP Client . Server based on Linux OS Ubuntu

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    Each file is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Vietnamese, you must also indicate the dialect) accor...rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Vietnamese, then it is necessary to indicate “Vietnamese (dialect name)”; c) if in addition to Romanian there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Vietnamese (dialect), spanish*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    Users are not able to make calls to some countries currently. The calls are not even getting to our switch (provider for international calls). We need to check what is wrong e.g if it is a setting or something else that needs to be updated.

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    Kamailio with CPS S'est terminé left

    Currently, Customer has a few dialers using asterisk and sending call to a single provider. Combined call can go can go beyond 40 calls per second, but provider only provide 30 CPS (Calls per seconds) We currently have 2 providers (A) and (B) We need Kamailio to first group the 30 calls and send them to Provider (A) every second, and overflow / remaining within the same second to send to Provider (B) Priority will go to (A) and than (B) callflow; 1. collect and gather all the calls from multiple dialer. 2. to control and send calls in batches, 30 cps. (user define number) 3. if call exceed 30 cps within the same second, kamailio will re-route next 30 calls to another provider (B) 4. able to do media bypass Thanks for reading

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    We have a product that consists of a desktop client (at student side), and a web application in Django (instructor side). We have few features that are messaging type or status type. For example, we need the following: - to have a direct chat between student and instructor - broadcast message from instructor to all students - If the student is online or offline (close the application) - And some other status updates We are currently working in polling mode. Each clients sends a request message to the server every 4 seconds and gets the available information about these details from database. This of course causes the database to be overloaded. We would like to redesign this part. We want a consultant to help us in choosing the correct technology or service after understanding our system W...

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    ...the billing portal through and they format of displaying client billing information is subject to the requirements 3) PBX portal - This is where the real work happens, building the pbx requires your knowledge in running native base code such as Golang to create api with connecting feature to the pbx system. The pbx module would be the pbx system dialplan design using asterisk and it should generate dialplan base on the asterisk code when that front end on the self service request configuration from the pbx. 4) Payment gateway - The portal must facilitate with payment gateway and it allows clients to see payment history on a click away. The historical data consist of their subscription and call usage where they can refer to instantly and this payment which we see should b...

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    Make Asterisk Dialplan S'est terminé left

    need to make dialplan for realtime queue using mysql for asterisk system

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    Hi Arthur N., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    I need a asterisk developer S'est terminé left

    Fusionpbx integrations and customisations

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    + You must have very good experience with Customizing Asterisk / Vicidial according to requirements + Ideally coding experience is desirable + Several Asterisk / vicidial Customizations to be completed

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    There are 163 entries in total. Each duration is no longer than 1 minute. The assignment requires: 1. Indicate the gender of all speakers on the recording; 2. Indicate all the languages ​​represented on the record (for Italian, you mus...following rules: a) if the language is not familiar, then indicate “unknown. language.”; b) if the language is only Italian, then it is necessary to indicate “Italian (dialect name)”; c) if in addition to Romanian there is some other language (1-2 words are not a full phrase), then you must specify the name of the second language with an asterisk. Example: “Italian (dialect), spanish*”; d) if there are full-fledged phrases in another language, then an asterisk next to the second language does not need to be...

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    ...build a proof of concept Java multithreaded standalone application that will connect to one or multiple Asterisk servers using AMI protocol (preferably using a library) and trigger a callback function when receiving events. At the same time, one or multiple clients can connect to this Java application using a predefined port (like 5039) and using the AMI protocol receive the events from the asterisk servers and send commands to one or multiple clients. When an event is received from the asterisk server, the application will send the same event to all or some of the clients connected. When a command is received from any of the clients connected, the same command will be sent to the asterisk server. In other words, I need a "wrapper". This is just a p...

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    I need to setup Nagios or similar monitoring software to monitor my network. The Nagios server will be hosted in AWS and I will need to monitor systems (Windows & Linux) on the internet and from behind NAT. -servers -apps (Asterisk, Apache, MySQL etc) -Network outages I need someone who is familiar with this type of work and will provide me with better ways to accomplish this is possible.

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    Call forward via API on asterisk S'est terminé left

    Hi Arshad N., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

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    ONLY for skilled DEV in Asterisk n Freepbx ######## Help in FreePBX/Asterisk CDR REPORTS Please check ScreenShots (ivr, logs, cdr) ######## Im realy BLIND in CDR reports about: where pbx is calling to (what number), where caller is going to (what IVR/Destination 1 or 2) All outgoing calls from Dongle1 are pointed to IVR, when someone pickup call it goes to IVR ("Message: Please choose 1 or 2"). ######## TASKS: A. DID number on CDR report - need to know where (what number) pbx is calling to B. DESTINATION - need to know what button/destination/extension caller goes (#1 or #2) this is tricky couse AT THE MOMENT i can see in DESTINATION number 9 (when user press1) and number 10 when user press2 C. LOG UNANSWERED CALLS in CDR report ######## IMPORTANT: Im usi...

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    ERPNext VOIP Integration S'est terminé left

    Wish to have an integration of an Asterisk PBX like FreePBX or something with ERP Next. We should have an incoming call popup with data being picked up from the ERP Next CRM Module for an existing number and We should have an option to save an unknown new number as a new Lead or Save to an existing customer/supplier / another contact similarly, we should be able to do a click-to dial from within ERP Next

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    VoIP Softswitch DID billing S'est terminé left

    We are currently looking for a new Softswitch, especially for DID Management / Billing. Instead of reinventing the wheel, we thought we would put feelers out there to see if maybe someone has already built one that a project has been canceled, that may be of use to us. We have very specific requirements and are happy to modify a current system. If not, we are looking at creating ...Management / Billing. Instead of reinventing the wheel, we thought we would put feelers out there to see if maybe someone has already built one that a project has been canceled, that may be of use to us. We have very specific requirements and are happy to modify a current system. If not, we are looking at creating one from scratch. We prefer if the project used Freeswitch / Kamailio instead of an ast...

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    À la une
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    Backend Python Developer S'est terminé left

    We are looking for an experienced python developer that will be responsible for creating a Python basked Backend API for a Telecommunications platform. You can think of it as Twilio + Vonage. This platform will be used to provide telecom functionality for our customers. We use MySQL (Tortoise ORM) as our backend database, Asterisk for our Media and RTP traffic, Kamailio for our front end Registration Servers. You will also be responsible for creating a client side API to access PBX features. This API will be used by our front end interface for customers and also will have Admin access for our internal CRM system to provision and manage customers.

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    Jak w tytule, potrzebuję zmodyfikować kod źreódłowy softu Asterisk (C / C++) żeby: - wysyłał dźwięk do zewnętrznego API (Google Api Speech-to-text) - na bieząco odczytywał rezultat z tego API i umieszczał wynik w zmiennej Asteriska - przykłady bibliotek Google API są umieszczone na Google Cloud Docs

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    I would need to Integrate Vtiger CRM 7.X and Issabel PBX. functions NEEDED on the Vtiger interface: - po...Issabel PBX. functions NEEDED on the Vtiger interface: - popup with caller information (only for the agent who answers) - click2call - History (including recording) What will I provide: - a fully working VPS - vtiger and Issabel (Asterisk) already installed on the VPS Important additiona informatins: - All the extensions MUST be PJSIP - I have a working version of the integration between vtiger and Issabel, but it doesn't work with PJSIP extensions. In case you are interested into working on this version, just let me know! - If you plan to customize the Asterisk callflow, you have to warn me PS: For those who don't know issabel, it is a Freepbx fork. ...

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    Hello Ervin, I would like to ask you if you may be interested in a project like this: I would need to Integrate Vtiger CRM 7.X and Issabel PBX. With the following functions: - popup on vtiger dashboard when the agent answer - click2call - History (including recording) Vtiger and Issabel are already installed and configured by us on the same VPS. Only integration would be needed Thanks in advance Andrea

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    This project is about "ISP CRM" With freeradius, Asterisk and Network Monitoring Capability which Support Vendors [Mikrotik, Huawei, Cisco, Juniper and Star OS]. Required Features and Modules # Admin Control Panel (Web + Android ) * User Management * Plan/Tariff Management * Finance Management >Invoice generation and payment processing >Recurring billing engine >Prepaid billing engine >Data Cap and Top UPs >Voice billing (VoIP CDR auto processing) >Payment gateways and credit card processing (India). >Cash Desk module * Accounting software integration (QuickBooks and Tally) * Referral Management * Sales Management and Onboard Process * Support Management * Task Management * ...

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    Vicidial (Asterisk) Customization S'est terminé left

    + You must have very good experience with Customizing Asterisk / Vicidial according to requirements + Ideally coding experience is desirable + Several Asterisk / vicidial Customizations to be completed Before hiring you to our long term projects, there will be a small test which needs to be cleared. Please share screenshots, once you are successful with the test. Test Use REST API, collect all Leads with below fields, from and add display it in your Django app. Username - Demo Password - Demo (For MD5 password, you can translate ‘Demo’ online) REST URL endpoint: Module name for connecting to SuiteCRM Leads Field names to collect are ‘ phone_work, first_name,

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