Register asterisk to sip trunk

Solution 1:

The config looks fine at first sight. Go on and try to debug your setup:

1. use "sip show registry" inside of asterisk to display the ougoing registrations
2. enable sip debugging: "sip set debug on" (shows the sip traffic within asterisk cli)
3. force a register attempt: "sip reload" and monitor the cli for appearing sip messages

If step 2 only shows outgoing but not incoming packets, you might have a firewall issue.

Solution 2:

First, the register line should have a path set at the end, like:

register => myusername:mypassword:[email protected]/84106639

Then do a sip reload or service asterisk restart. After that, the sip show peers command should return some kind of status. The qualify=yes option is useful too to check IP connectivity and SIP service status.

Also, your [outgoing] rule in the dialplan should be:

exten=> _03.,1,Dial(SIP/flowroute/${EXTEN},30)

Solution 3:

You are missing two items to be placed directly under your register => entry.

[general]
register => myusername:[email protected]
registertimeout=20
registerattempts=0

As soon as you enter these in sip.conf, at the asterisk console enter:
*CLI> sip reload
*CLI> sip show registry

Solution 4:

i) use qualify=yes in sip.conf then sip reload and run command sip show peers and check status is ok or not of your flowroute.

ii) Also stop firewall service and check status.