Strange issue with calling phone A->B ; B->A

Asterisk Asterisk Support
1

Hallo… thats asterisk 1.8.2.2
Behind NAT
simple SIP simple extensions, with mysql backend.
I hope someone could help.
I CAN call from phone A to phone B
but its impossible to call from phone B to phone A
CLI:

Phone A -> Phone B

== Using SIP RTP CoS mark 5

-- Executing [000001@default:1] Dial("SIP/000000-00000076", "SIP/000001") in new stack

== Using SIP RTP CoS mark 5

-- Called 000001

-- SIP/000001-00000077 is ringing

-- SIP/000001-00000077 answered SIP/000000-00000076

-- Locally bridging SIP/000000-00000076 and SIP/000001-00000077

== Spawn extension (default, 000001, 1) exited non-zero on ‘SIP/000000-00000076’

Phone B -> phone A

== Using SIP RTP CoS mark 5

-- Executing [000000@default:1] Dial("SIP/000001-00000078", "SIP/000000") in new stack

[Jul 12 19:08:35] WARNING[2965]: app_dial.c:2039 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)

== Everyone is busy/congested at this time (1:0/0/1)

-- Executing [000000@default:2] Hangup("SIP/000001-00000078", "") in new stack

== Spawn extension (default, 000000, 2) exited non-zero on ‘SIP/000001-00000078’

Any sugestions?

2

Channel unavailable. There is no 000000 in sip.conf, or it has a dynamic address and the phone hasn’t registered.

Warning, the fact that it works the other way may mean you have allowguest=yes. That is inadvisable unless you know what you are doing.

3

Hmmm
000000 is in sip (MySQL)

000001 is registering

– Registered SIP ‘000001’ at 62.85.93.44:59747
> Saved useragent “CSipSimple r977 / GT-I9000-10” for peer 000001

000000 is registering too

– Registered SIP ‘000000’ at 62.85.93.44:64686
> Saved useragent “3CXPhone 6.0.19548.0” for peer 000000

4

The issue was a Firewall on client side.