Can't dial out

Asterisk Asterisk Support
1

I’m trying to dial out of my server and I am getting the following errors.

== Console is full duplex
– Executing [15196417193@default:1] Dial(“Console/dsp”, “SIP/lesnet/15196417193|60”) in new stack
Audio is at xx.xx.xx.xx port 17258
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.34.181.47:5060:
INVITE sip:15196417193@64.34.181.47 SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK494822c8;rport
From: “asterisk” sip:asterisk@xx.xx.xx.xx;tag=as23556e5a
To: sip:15196417193@64.34.181.47
Contact: sip:asterisk@xx.xx.xx.xx
Call-ID: 5b2797f20f64dd7a56936f09413d4a3a@xx.xx.xx.xx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 21 May 2008 22:05:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 17440 17440 IN IP4 xx.xx.xx.xx
s=session
c=IN IP4 xx.xx.xx.xx
t=0 0
m=audio 17258 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called lesnet/15196417193

asterisk*CLI>
<— SIP read from 64.34.181.47:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK494822c8;received=206.248.190.155;rport=5060
From: “asterisk” sip:asterisk@xx.xx.xx.xx;tag=as23556e5a
To: sip:15196417193@64.34.181.47;tag=as71bf35f3
Call-ID: 5b2797f20f64dd7a56936f09413d4a3a@xx.xx.xx.xx
CSeq: 102 INVITE
User-Agent: LES.NET.VoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Transmitting (no NAT) to 64.34.181.47:5060:
ACK sip:15196417193@64.34.181.47 SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK494822c8;rport
From: “asterisk” sip:asterisk@xx.xx.xx.xx;tag=as23556e5a
To: sip:15196417193@64.34.181.47;tag=as71bf35f3
Contact: sip:asterisk@xx.xx.xx.xx
Call-ID: 5b2797f20f64dd7a56936f09413d4a3a@xx.xx.xx.xx
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

-- SIP/lesnet-081df1d8 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)
– Executing [15196417193@default:2] PlayTones(“Console/dsp”, “congestion”) in new stack
– Executing [15196417193@default:3] Congestion(“Console/dsp”, “”) in new stack
Really destroying SIP dialog '5b2797f20f64dd7a56936f09413d4a3a@xx.xx.xx.xx’ Method: INVITE

2

my configs:

sip.conf

[general]
context=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)

disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=alaw ; Allow codecs in order of preference
;useragent=Fullmotions.com ; Allows you to change the user agent string
;dtmfmode = auto ; Set default dtmfmode for sending DTMF. Default: rfc2833

;externip=xx.xx.xx.xx ; Address that we’re going to put in outbound SIP
;localnet=192.168.0.0/255.255.255.0; All RFC 1918 addresses are local networks
defaultip=xx.xx.xx.xx

[lesnet]
type=friend
host=did.voip.les.net
dtmfmode=rfc2833
insecure=port
disallow=all
allow=ulaw
context=default

register => 8998:1234@192.168.0.183
[8998]
type=peer
host=dynamic
username=8998
secret=1234
disallow=all
allow=ulaw
context=default

register => 8999:1234@192.168.0.200
[8999]
type=peer
host=dynamic
username=8999
secret=1234
disallow=all
allow=ulaw
context=default

extensions.conf

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Zap/1

[default]
exten => 8999,1,Dial(SIP/8999,20) ; permit transfer
exten => 8999,n,Voicemail(8999,u) ; Voicemail (unavailable)
exten => 8999,n,HangUp()

exten => 89981,1,Dial(SIP/8998,20)
exten => 89981,n,Voicemail(8999,u)
exten => 89981,n,HangUp()

include => outgoing;

[outgoing]
exten => _1NXXNXXXXXX,1,Dial(SIP/lesnet/${EXTEN})
exten => _1NXXNXXXXXX,n,Hangup()
exten => _1NXXNXXXXXX,102,busy()

exten => _NXXNXXXXXX,1,Dial(SIP/lesnet/1${EXTEN})
exten => _NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,102,busy()

3

There is a 404 error in there. It could be you are not sending the number in the correct format. It can also be an indication of incorrect authentication parameters. Speak to your ITSP and ask them what they are getting on their end.

4

Well what is happening is I can dial sometimes, most of the time I can’t. Sometimes it looks like it is dial out on my local IP address and not through the lesnet peer. I’m not sure what it is. I’ve tried a lot of different configurations. I spent some time on the phone with my provider and they said they did not see any traffic from my server hitting theirs.

Thanks.

5

What do you mean by “. Sometimes it looks like it is dial out on my local IP address and not through the lesnet peer.”. Seems to be more of a Linux issue (maybe firewall) or maybe an issue with your router.