VoIP Voice Over IP
Linux
In this article you will learn how to install TURN Server on Ubuntu 20.04 LTC for WebRTC, configure Coturn with Long Term Credential Mechanism, configure IPTables firewall, check TURN server. The TURN server implements the STUN protocol also. Coturn is free open source TURN server. Requirements To install Coturn
FreeSWITCH
In this article I will how to fix FusionPBX wrong time. FreeSwitch time can be drift from the Linux system time. It will create problem for FusionPBX CDR and Time condition functionality. Lets compare Linux System time and FreeSwitch time To display the current System Time in Linux type: $ date
FreeSWITCH
How to fix FusionPBX problem when unable start Internal Profile – internal: Invalid Profile [internal]. In debug logs: Debug shows: [ERR] switch_core_db.c:108 SQL ERR [file is encrypted or is not a database] Remove .db files that are in the DB folder inside your FreeSwitch install # cd /var/
FreeSWITCH
In this article I will show how to enable multiple registrations on FusionPBX. It will help have multiple VoIP phones registered in FusionPBX or FreeSwitch, and receive simultaneous incoming calls on that devices. Multiple registrations would need to be enabled in the SIP Profile. This would enable the 2 to
FreeSWITCH
In this article you will learn how to install FreeSWITCH G729 Free Codec on FusionPBX. Time needed: 10 minutes. How to install G729 Free Codec on FreeSWITCH 1. Download free g729 codec Download mod_bcg729.so file from http://repo.sip247.com/debian/ 2. Put g729 codec into FreeSWITCH modules
FreeSWITCH
In this article you will learn how to convert FusionPBX Call Recordings from WAV to OGG (OPUS) with Linux script. This script finds yesterday *.wav FisionPBX call recordings, convert them to opus and then delete. After updates CDR records. Script converts files with FFmpeg converter. And should executed daily with
Asterisk
Asterisk SIP Trunk Registration Example Below you see Asterisk SIP trunk registration simple example. So edit sip.conf file and add register string to register Asterisk SIP trunk in [general] section. register=>SIP_ID:Password@Your_SIP_Provider_Host_or_IP Then fill host, username, secret with your SIP
Asterisk
Using tcpdump for SIP and RTP Diagnostic (tcpdump examples) The tcpdump is a powerful command-line packet analyzer may use for for dumping traffic on a network. For more information about tcpdump options see man pages. Below you see tcpdump command examples filter for diagnostic VoIP SIP Install tcpdump To install
VoIP
Adtran Total Access 924 Initial Configuration Connect to CRAFT port. There is no username password by default. TA924> en TA924# TA924# configure terminal TA924(config)# interface eth 0/1 TA924(config-eth 0/1)# no shutdown TA924(config-eth 0/1)# ip address dhcp TA924(config-eth 0/1)# end Check the
Asterisk
List of SIP response codes 1xx—Provisional Responses 100 Trying Extended search being performed may take a significant time so a forking proxy must send a 100 Trying response. 180 Ringing Destination user agent received INVITE, and is alerting user of call. 181 Call is Being Forwarded Servers can optionally
FreeSWITCH
FusionPBX Rejecting Incoming Calls From SIP Trunk Usually fresh installed FusionPBX Rejecting Incoming Calls From SIP Trunk by default. To fix problem you need add carrier to the domains acl set the node CIDR to allow. Like ProviderIP/32 under CIDR. Goto menu Advanced -> Access Control edit “domains”
VoIP
T38 FAX. Cisco SPA122 Configuration How to enable T38 FAX support on Cisco SPA122 VoIP adapters. Proxy and Registration In “Proxy and Registration” and “Subscriber Information” fill Proxy: YourSIP.Server User ID: SIP Username Password: SIP Password NAT Settings Goto Voice –> Line 1 or Line 2 depending your lines