Troubleshooting SIP
One of the things that Asterisk is not good at is providing good simple troubleshooting information (at the time of this writing - Dec 08). I am sure that in the next year, many great tools will pop up to simplify troubleshooting problems. If you have no SIP help menus showing at the Asterisk CLI, it may be that CentOS is recognizing your ethernet card as a device other than eth0.
This webpage is Part One in troubleshooting a SIP trunk between two Asterisk PBXs. The information can be used to troubleshoot a SIP extension problem also.
The initial setup is PBX 187 (10.163.110.187) trunking to PBX 136 (10.163.110.187) using SIP.
Here is the SIP trunk configuration from FreePBX for each:
Here's a list of Asterisk CLI commands that will aid in determining the problem:
- "sip show peers" indicates 136-SIPpeer/187-SIPpeer.
- 136-SIPpeer is the Server to connect to
- 187-SIPpeer is the username.
- The OK indicates that you can ping to it - not that it is working properly!
- "sip show channels" will show all of the SIP channels currently registered:
- "sip show registry" will show all of the SIP extensions currently registered:
- "sip show users" will show all of the SIP users currently registered:
The Problem
When you use a phone connected to PBX 187, dial 6 and the extension 2001 to get to PBX 136, the dreaded message "All circuits are busy now. Please try your call later." appears. Extension 2001 has been verified to work properly at PBX 136. Next step is to go to Part Two: Identifying the problem.
