The History of VoIP Voice over Internet Protocol, abbreviated VoIP, is a fairly recent technology. It goes by many names, including Voice Over Broadband (VoBB), Internet Telephony, IP Telephony, and broadband phone.The advent of the Internet brought with it an abundance of innovation regarding communication. Of...
Even though we have used it in the past, we would not recommend it for professional use. However, please try the below configuration (update the bold values): 1. Asterisk Side SIP TRUNK Trunk Description: PSTN Outbound Caller ID: Your Phone Number Maximum Channels: 1 Outbound Dial Prefix: 9 (Leave it blank if the...
Elastix – PBX: 1.Select – Trunks – Add Sip Trunk General Settings Trunk Description: PSTN Outbound Caller ID: YOUR PHONE NUMBER CID Options: Allow any CID Maximum Channels: 1 Outgoing Settings Trunk Name: PSTN PEER Details context=from-trunk username=300 (USE THE SAME ID ON YOUR GW) secret= YOUR PASSWORD (USE THE SAME PASSWORD ON YOUR GW) host=dynamic port=5061 srvlookup=yes insecure=port,invite canreinvite=no dtmfmode=rfc2833 t38pt_udptl=yes nat=no qualify=yes type=peer disallow=all allow=alaw&ulaw sendrpid=yes Submit and apply! 2.Select Outbound...
Which Cisco 79??. It could be a later model, which you could flash it so as to use SIP. Anyway, try the below: 1. Download and install sccp, Chan_SCCP. cd /usr/src wget https://downloads.sourceforge.net/project/chan-sccp-b/V3/Chan_SCCP-3.0_RC3.tar.gz tar xvf Chan_SCCP-3.0_RC3.tar.gz cd Chan_SCCP-3.0_RC3 ./Configure make make install 2.If there is no TFTP server on your pbx, please do install (it exists on elastix). 3.Edit and...
The NAT configuration can be found in /etc/asterisk/sip_general_custom.conf (on elastix, at least) and not in sip.conf as stated in other forums. Do not edit sip.conf. The example below assumes that your Asterisk PBX has an IP address of 192.168.1.X: externip=XXX.XXX.XXX.XXX (This is your static public IP. Instead of...