Tools for covering the documentation of Asterisk Dialplan, including contexts, variables, etc. Including having a proper flow..
I have a problem for received calls form my Grandstream HT-503. I have a FXO connect to my PABX, and I can make a call from PABX to VOIP, but I didnt received calls to my VOIP, t..
Im facing strange issue while establishing inbound calls from SIP trunks. Provider A is sending (G729, Alaw, uLaw) offer and asterisk dial the peer with its preferred codec order(G729, aLaw, uLaw). The peers phone send the codec list as (uLaw, spe..
Some background information.I have used Debian with Asterisk for several years. Have encountered zero problems. I am now trying to setup an Asterisk on a CentOS7 box using VMWare Workstation. I am brand new to CentOS and RHEL so I may be missing someth..
After all these years installing from source, Im giving Dahdi package installation a try on a recent Stretch box.Note: Sangoma’s USBfxo is a low-cost development tool kit, ideal for those just getting started with Asterisk® or those who just w..
I have webrtc up and running with asterisk 11. All is going well with TLS now working. At least I hope it is using TLS and wss. Based on what I am seeing I have UDP, WSS listed in the Allowed transports, but every time I connect the Primary transp..
I have a setup which is not working right now:Provider – DSL-Router (192.168.2.1) – Bintec-Router (10.17.46.66) – Asterisk (10.17.46.99)Note: upgrade your network to wireless AC for reliable, high-speed Wi-Fi up to 1200Mbps. Enjoy more HD gaming ..
The error Unable to create channel of type SIP (cause 20 – Subscriber absent) usually means that the SIP device hasnt regis..