Sip trunk host=dynamic

Asterisk Asterisk Support
1

Hi Guys

i´ve a problem with my system, i hope you can help me.
I try to connect 2 Asterisk by a sip trunk. If i look at sip show peers and sip show registry everthing is ok.
This is my config:

Server2:
[xxx]
username=xxx
type=friend
secret=kjf72356kfh5
host="ip"
disallow=all
allow=alaw
allow=ulaw
dtmfmode=rfc2833
context=default

register=xxx:kjf72356kfh5@ip/xxx

server2:
[xxx]
username=xxx
type=friend
secret=kjf72356kfh5
host=dynamic
disallow=all
allow=alaw
allow=ulaw
dtmfmode=rfc2833
context=from-test-outbound

Calls from server1 to server2 work, but calls from sip1 to com1 doesn´t work.
A console dial from sserver2 tells me this:

– Executing [1100@from-pstn:8] Dial(“Console/default”, “SIP/123456789/1100”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [1100@from-pstn:9] NoOp(“Console/default”, “es geht weiter”) in new stack
– Executing [1100@from-pstn:10] Hangup(“Console/default”, “”) in new stack

Over a tcpdump i can see that server2 doesn´t send anything to com1.
If i change the host=dynamic from server2 to the host=“ip” it works.
But the host need to be dynamic

Hope you can help me.

Thank you for your help.

MotherTrucker

2

Has nobody any Idea or something else?

3

You haven’t provided enough information. You’ve even not recorded the message reporting the peer as unavailable.

Most likely, the registration failed.

Normally, in this sort of case, one bypasses the problem by making both sides have static addresses.

Note, you shouldn’t use the same name for both peers, although I’m not convinced that will actually cause the problem.

Also, you should use type=peer, although, again, that is probably not the problem here.