Asterisk debugging

This page has tips on what to do when debugging Asterisk problems

One way audio

Typically happens when remote phone and server are NATTED (one end or both)

Make sure remote phone is set to nat=? (where ? != no) so that Asterisk ignores the non-routable address in the SIP header and uses the packet source instead.

There has to be a clear path both directions for the RTP stream, which is where the media (audio) is transported. That means the SIP session can run just fine but if the RTP stream is blocked in one direction, the recipient of that stream will get no audio channel.

Since RTP flows over a UDP port there is no built in error correction, the sender just blindly spits out UDP packets without knowing if they are getting through or not. This makes it a little more mysterious.

Asterisk Variable

http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List

MySQL

When using MySQL you can turn on MySQL logging and watch the queries

SIP

When debugging calls you can turn on SIP debugging like this

sip set debug on

CDS Mikrotik router

Rules to enable Vastra (NUC)

add action=dst-nat chain=dstnat comment="Vastra" dst-address=107.135.132.129 dst-port=5060 protocol=udp to-addresses=192.168.2.234 to-ports=5060 add action=dst-nat chain=dstnat comment="Vastra RTP audio stream" dst-address=107.135.132.129 dst-port=10000-19999 protocol=udp to-addresses=192.168.2.234 to-ports=10000-19999

Rules to enable Vastra2

add action=dst-nat chain=dstnat comment="Vastra2 PBX" dst-address=107.135.132.129 dst-port=5060 protocol=udp to-addresses=192.168.2.235 to-ports=5060

add action=dst-nat chain=dstnat comment="Vastra2 RTP audio stream" dst-address=107.135.132.129 dst-port=10000-19999protocol=udp to-addresses=192.168.2.235 to-ports=10000-19999

Rules to MODIFY

set 0 to-addresses=192.168.2.235