Outgoing calls

Asterisk Asterisk Support
1

Hi All.

I have just installed asterisk 1.4.20. When I am dialing out I am getting “The number you have dialed…”

Here is my config and output:

exten => _9XXXXXX,1,Dial(Zap/g1/${EXTEN})

– Executing [9116696@internal:1] Dial(“SIP/2000-082155b0”, “Zap/g1/9116696”) in new stack
– Requested transfer capability: 0x00 - SPEECH
– Called g1/9116696
– Zap/1-1 is proceeding passing it to SIP/2000-082155b0
– Hungup ‘Zap/1-1’
== Spawn extension (internal, 9116696, 1) exited non-zero on ‘SIP/2000-082155b0’

2

Ok, have ran a debug so here is some more information:

<------------>
– Executing [9111696@internal:1] Dial(“SIP/2000-08210dc0”, “Zap/g1/111696”) in new stack
– Requested transfer capability: 0x00 - SPEECH
– Called g1/111696
Audio is at 10.6.0.58 port 11628
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 10.6.0.125:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.6.0.125:5060;branch=z9hG4bK-cba7879f;received=10.6.0.125
From: “Chris Owen” sip:2000@10.6.0.58;tag=71358c37eccac72fo0
To: sip:9111696@10.6.0.58;tag=as58604a71
Call-ID: 6ac416d7-5bbcbd0f@10.6.0.125
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:9111696@10.6.0.58
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 5052 5052 IN IP4 10.6.0.58
s=session
c=IN IP4 10.6.0.58
t=0 0
m=audio 11628 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

<------------>
– Zap/1-1 is proceeding passing it to SIP/2000-08210dc0

<— SIP read from 10.6.0.125:5060 —>
CANCEL sip:9111696@10.6.0.58 SIP/2.0
Via: SIP/2.0/UDP 10.6.0.125:5060;branch=z9hG4bK-cba7879f
From: “Chris Owen” sip:2000@10.6.0.58;tag=71358c37eccac72fo0
To: sip:9111696@10.6.0.58
Call-ID: 6ac416d7-5bbcbd0f@10.6.0.125
CSeq: 102 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest username=“2000”,realm=“asterisk”,nonce=“1e742e39”,uri="sip:9111696@10.6.0.58",algorithm=MD5,response="e20294411662d9f9918a222c4c4bd261"
User-Agent: Linksys/SPA942-5.2.8
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 10.6.0.125 : 5060 (no NAT)

<— Reliably Transmitting (no NAT) to 10.6.0.125:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.6.0.125:5060;branch=z9hG4bK-cba7879f;received=10.6.0.125
From: “Chris Owen” sip:2000@10.6.0.58;tag=71358c37eccac72fo0
To: sip:9111696@10.6.0.58;tag=as58604a71
Call-ID: 6ac416d7-5bbcbd0f@10.6.0.125
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>

<— Transmitting (no NAT) to 10.6.0.125:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.6.0.125:5060;branch=z9hG4bK-cba7879f;received=10.6.0.125
From: “Chris Owen” sip:2000@10.6.0.58;tag=71358c37eccac72fo0
To: sip:9111696@10.6.0.58;tag=as58604a71
Call-ID: 6ac416d7-5bbcbd0f@10.6.0.125
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:9111696@10.6.0.58
Content-Length: 0

<------------>
– Hungup ‘Zap/1-1’
== Spawn extension (internal, 9111696, 1) exited non-zero on ‘SIP/2000-08210dc0’

3

ok,

Fixed this by add in /etc/asterisk/zapata.conf

pridialplan=unknown
prilocaldialplan=unknown

Thanks