VoIP Voice Over IP
VoIP
Configure Distinctive Ring on Grandstream For set up Distinctive Ring on Grandstream GXP2130, 2140, 2160 phones Goto: Accounts -> Select the account (1,2,3 or 4) -> Call Settings In section Match Incoming Caller ID fill Matching Rule fields For example: * xxxx – Will work with 4 digits
Cisco
How to fix Cisco SPA504G Time In “Regional” tab Set: Ignore DHCP Time Offset: yes Daylight Saving Time Enable: yes Daylight Saving Time Rule: start=10/1/7/03:0:0;end=4/1/7/02:0:0;save=1
VoIP
How To Enable Web GUI on Adtran Totall Access TA 924 from console, and configure static IP on Ethernet port First you need connect your Adtran TA to your PC via console port. Also make sure your computer and Adtran connected to same network switch and set on your computer
Linux
How to capture SIP packets with tcpdump (example) tcpdump -nt -s 0 -A -i eth0 port 5060 -w call.cap Tcpdump will create file call.cap Capture DHCP requests with tcpdump
FreeSWITCH
Verto (VER-to) RTC is a FreeSWITCH endpoint that implements a subset of a JSON-RPC connection designed for use over secure websockets. This allows a web browser or other WebRTC client to originate a call using Verto into a FreeSWITCH installation and then out to the PSTN using SIP, SS7, or
FreeSWITCH
You must install libldns-dev to build mod_enum During the install FreeSWITCH on CentOS (Centos/RHEL 6.4 and 6.5, 6.6) you may getting an error: “You must install libldns-dev to build mod_enum. Stop.” yum install libldns-dev can’t help: No package libldns-dev available Error: Nothing to
Asterisk
Elastix admin password reset For Elastix admin password reset just execute from Linux terminal: # sqlite3 /var/www/db/acl.db "update acl_user set md5_password='7a5210c173ea40c03205a5de7dcd4cb0' where id=1" You will be able login to WEB GUI with new password: palosanto
sipXecs
sipXecs port forwarding Sometimes you may have one way audio (one way RTP) problem on sipXecs phones. Correct functioning sipXecs registered phones behind of NAT requires ports forwarding. Or if network has firewall next ports should be opened: 5060 TCP and UDP 5080 TCP and UDP 10000 - 31000 UDP
Asterisk
How to Configure Motorola Netopia 3000 port forwarding, SIP ALG First we need configure Port forwarding: Login to WEB GUI: http://192.168.1.254 Configure -> Advanced -> Pinholes for SIP UDP transport Protocol: UDP External: Port Start 5060 External: Port Start 5080 Internal: IP Address: Internal:
Asterisk
To start Asterisk at boot time, do the following from Linux prompt: chkconfig --add asterisk