Alternatives and similar repositories for gofaxip

Users that are interested in gofaxip are comparing it to the libraries listed below

• provisioner / Provisioner
  The is the central location for the Provisioner Module for VoIP/PBX Servers. Most of the new work is happening inside the v5-dev branch
  ☆123Updated last year
• palner / apiban
  REST API for sharing IP addresses sending unwanted SIP traffic
  ☆62Updated last year
• jpawlowski / freeswitch-sounds-tts
  FreeSWITCH TTS Voice Prompt Generator
  ☆43Updated 6 years ago
• voiceboys / sbcOS
  Barebone Opensource Powered SBC
  ☆109Updated 10 months ago
• traud / asterisk-opus
  Asterisk 13 transcoding module: Opus
  ☆32Updated 3 years ago
• surendratiwari3 / KamInboundSIP
  This Project will provide the inbound sip using that we can route did call to customer ip or customer Phone number
  ☆24Updated 7 years ago
• seanbright / asterisk-opus
  Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration
  ☆35Updated 8 years ago
• dOpensource / dsiprouter
  UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services
  ☆210Updated this week
• jchavanton / voip_patrol
  VoIP signaling and media test automation
  ☆119Updated last month
• ghrst / Kamailio-HA
  This repository contains Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.
  ☆54Updated 6 years ago
• etamme / federated-sip
  Federated SIP deployment
  ☆35Updated last year
• asipto / siremis-php
  Web Admin Interface for Kamailio
  ☆103Updated 4 months ago
• blox-org / blox-cfg
  Blox route configuration (opensips script)
  ☆22Updated 4 years ago
• ossobv / asterisklint
  Asterisk PBX configuration syntax checker
  ☆66Updated 2 years ago
• meetecho / asterisk-opus
  Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration
  ☆80Updated 10 years ago
• negbie / heplify-xrcollector
  ☆9Updated 6 years ago
• iNextrix / ASTPP
  Open Source VoIP Billing Solution http://www.astppbilling.org | Report a bug https://inextrix.atlassian.net/jira/projects
  ☆163Updated 3 weeks ago
• voxserv / freeswitch-perf-dialer
  FreeSWITCH dialer program for VoIP performance tests
  ☆46Updated 7 years ago
• sipcapture / sipgrep
  SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console
  ☆168Updated 3 weeks ago
• sipcapture / captagent
  100% Open-Source Packet Capture Agent for HEP
  ☆173Updated 2 months ago
• Viking-VoIP / full-voip-on-k8s
  Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCH
  ☆54Updated 2 years ago
• BelledonneCommunications / flexisip
  Linphone.org mirror for flexisip (git://git.linphone.org/flexisip.git)
  ☆164Updated this week
• urueedi / kazoo-provision
  automatic device provision in monster-ui-voip
  ☆19Updated 9 years ago
• retel-io / ari-proxy
  Ari-proxy connects Asterisk, an open source communication server, to the Apache Kafka distributed streaming platform.
  ☆61Updated last week
• voxserv / freeswitch_conf_minimal
  Minimalistic FreeSWITCH configuration as a start for new projects
  ☆82Updated 8 years ago
• gmaruzz / sipnagios
  Nagios Plugin to check Call Quality in SIP VoIP (compatible checkmk, etc)
  ☆31Updated 4 years ago
• asipto / secsipidx
  Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library
  ☆50Updated 6 months ago
• sipwise / kamailio-config-tests
  Kamailio configuration tests
  ☆10Updated last week
• nethesis / falconieri
  Remote Provisioning Gateway
  ☆19Updated last month
• pascomnet / asterisk_sbr
  Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.
  ☆44Updated 3 years ago