Dial() aplication doesn`t work

Asterisk Asterisk Support
1

Hello
I`m new in asterisk, and i have problem.
My asterisk is connected to voip provider via sip trunk, and i have inside client connected via x-lite.
X-lite client can call outside, and calls from aoutside are established to asterisk server.
Problem is when i call from outside to asterisk, and press 101 in order to forward call to inside user 101.
In console i sew then:

[Dec  8 14:21:28] WARNING[32161]: chan_sip.c:3005 create_addr: No such host: 101
[Dec  8 14:21:28] WARNING[32161]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)

Tcpdump then shows, that asterisk server is trying to resolv name 101 in dns server, so It everything seems to be problem with dial() apllication which is trying to forward call to outside. I don`t understund it, i thought that command

exten => 101,1,Dial(SIP/101) get call to user 101 definied in sip.conf.
Can Enybady explain it to Me, and tell wher is my problem?
My sip conf:

[general]
register => seipuser1:pass@sip-server.net/6929
register => sipuser2:pass@sip-server.net/6930
[authentication]
auth = sipuser1#pass@sip-server.net
auth = sipuser2#pass@sip-server.net

[frico1318]
type=peer
host=x.x.x.x
username=sipuser1
fromuser=sipuser1
secret=pass
context=begin
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=ulaw
insecure=invite
canreinvite=no
nat=no


[102]
type=user
host=dynamic
context=phones
secret=pass102
username=102
nat=yes

[101]
type=user
host=dynamic
context=phones
secret=pass101
username=101
nat=yesroot

extension.conf

[general]
autofallthrough=yes

[begin]
exten => 6929,1,Goto(6929,1,1)
exten => 6930,1,Goto(6930,1,1)

[6929]
exten => 1,1,Answer()
exten => 1,n,Background(welcome_1)
exten => 1,n,WaitExten(5)
exten => 101,1,Dial(SIP/101)
exten => i,1,Playback(pbx-invalid)
exten => i,n,Goto(6929,1,1)
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()

[6930]
exten => 1,1,Answer()
exten => 1,n,Background(welcome_przychodnia_4)
exten => 1,n,Hangup()

[phones]
exten => _X.,1,Answer()
exten => _X.,n,Dial(SIP/frico1318/${EXTEN})

Thanks for replies

2

101 is better to be friend, not user.
Users could dial out, peers could get calls and friends could do both.

3

thats right, i ganged it to friend. Dial still doesnt work properly. Console shows log:

  == CDR updated on SIP/sip-user-085af8e8
    -- Executing [101@6929:1] Dial("SIP/sip-user-085af8e8", "SIP/101") in new stack
[Dec  8 17:42:32] WARNING[2257]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)

Tcpdump doesnt try to resolv name 101 now, thats ok, but above logs shows that, dial() application tries forward call via sip trunk, not directly to sip user 101. Eny ideas ?

4

Can you show us the output for sip show peers in the CLI

5 
asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status          
101/101                    (Unspecified)    D   N      0        Unmonitored     
sip-user/sip-user    x.x.x.x              5060     Unmonitored     
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 1 offline]
6

As you can see your SIP device 101 isn’t registered, so the asterisk can’t dial or create the channel to dial.

You need to verify if your Device has the correct name,username and password; also verify if your firewall in your asterisk box is accepting connections to 5060 tcp port.

7

login and password are ok bacouse i am abble to make outbound connection using this user and call outside.
I don`t have firewall, all ports are allowed UDP and TCP in asterix box

8

With one unregistered Device you can make outgoings calls but no recieve. Run the command service iptables stop in your asterisk or /etc/init.d/iptables stop and then login in your CLI and restart your Device 101 you will see in the cli the registration of device 101.

And then try again, its important you see the cli when restart your device 101 to view if is registering correctly.

9

ok i solved it, thank you so much navaismo, your posts were very helpful.
The solution was setting proxy for incomming connection in x-lite

10

excellent! and you’re welcome
See you