Problems making outbound calls

I am very new to Freepbx. My buddy has helped me configure everything but it all went bad when I played with My DHCP server. Needless to say its all fixed now.

Trunk is online and registered, phone is registered, but every time I try to make an outbound call I continually get the “your call cannot be completed as dialed” error message.

as a newbie, any help at all would be appreciated

doesn’t look like you have an outbound route that will consume that number (per my previous comment).

Can you dial to an internal extension?
You may also want to make a new extension. I have found my own system, extensions refuse to function on 101 or 103

I think the developers just have too many conf an script files

Unfortunately I only have one phone at the moment.

Here is the output from the Asterisk CLI:
Connected to Asterisk 1.4.15 currently running on pbx (pid = 2686)
Verbosity is at least 3
== Manager ‘admin’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
– Executing [[email protected]:1] ResetCDR(“SIP/1000-09285640”, “”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/1000-09285640”, “”) in new stack
– Executing [[email protected]:3] Wait(“SIP/1000-09285640”, “1”) in new stack
– Executing [[email protected]:4] Playback(“SIP/1000-09285640”, “silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer”) in new stack
– <SIP/1000-09285640> Playing ‘silence/1’ (language ‘en’)
– <SIP/1000-09285640> Playing ‘cannot-complete-as-dialed’ (language ‘en’)
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
– <SIP/1000-09285640> Playing ‘check-number-dial-again’ (language ‘en’)
== Manager ‘admin’ logged off from 127.0.0.1
– Executing [[email protected]:5] Wait(“SIP/1000-09285640”, “1”) in new stack
– Executing [[email protected]:6] Congestion(“SIP/1000-09285640”, “20”) in new stack
== Spawn extension (from-internal, 19057412945, 6) exited non-zero on ‘SIP/1000-09285640’
– Executing [[email protected]:1] Macro(“SIP/1000-09285640”, “hangupcall”) in new stack
– Executing [[email protected]:1] ResetCDR(“SIP/1000-09285640”, “vw”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/1000-09285640”, “”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/1000-09285640”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [[email protected]:6] GotoIf(“SIP/1000-09285640”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/1000-09285640”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [[email protected]:11] Hangup(“SIP/1000-09285640”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/1000-09285640’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/1000-09285640’
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1

hopefully this will reveal the issue.

I think the spawn extension line has something to do with it.

Have you configured your outbound routes to direct the call to your trunk?

What number are you trying to dial and then what are the dialpatterns you have setup in your outbound routes?

You can download a free softphone such as X-Lite and have it connect to test.

What you posted does not tell us much.

“sip set debug” is better

“sip set debug off” when you are finished

Here is the Debug output:
pbx*CLI> sip set debug
SIP Debugging re-enabled
Reliably Transmitting (NAT) to 192.168.31.112:5060:
OPTIONS sip:[email protected];transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.31.31:5060;branch=z9hG4bK0bef65dd;rport
From: “Unknown” sip:[email protected];tag=as45cbbdb0
To: sip:[email protected];transport=udp
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 30 Aug 2009 17:10:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


pbx*CLI>
<— SIP read from 192.168.31.112:5060 —>
SIP/2.0 200 OK
Call-ID: [email protected]
CSeq: 102 OPTIONS
From: “Unknown” sip:[email protected];tag=as45cbbdb0
To: sip:[email protected];tag=f10f2fc5bf5a826
Via: SIP/2.0/UDP 192.168.31.31:5060;branch=z9hG4bK0bef65dd;rport
Content-Length: 0
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Contact: sip:[email protected];transport=udp
Supported: replaces
User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
pbx*CLI>
<— SIP read from 192.168.31.112:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.31.112;branch=z9hG4bKb1fd1d1b8
Max-Forwards: 70
Content-Length: 565
To: 19057412945 sip:[email protected]:5060
From: sip:[email protected]:5060;tag=38508459db22dfd
Call-ID: [email protected]
CSeq: 1806834509 INVITE
Supported: timer
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Content-Type: application/sdp
Contact: sip:[email protected];transport=udp
Supported: replaces
User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

v=0
o=MxSIP 0 1049955853 IN IP4 192.168.31.112
s=SIP Call
c=IN IP4 192.168.31.112
t=0 0
m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 BV16/8000
a=rtpmap:102 BV32/16000
a=rtpmap:107 L16/16000
a=rtpmap:104 PCMU/16000
a=rtpmap:105 PCMA/16000
a=rtpmap:106 L16/8000
=rtpmap:97 G726-16/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:99 G726-40/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=silenceSupp:on - - - -

<------------->
— (15 headers 23 lines) —
Sending to 192.168.31.112 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]

<— Reliably Transmitting (NAT) to 192.168.31.112:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.31.112;branch=z9hG4bKb1fd1d1b8;received=192.168.31.112
From: sip:[email protected]:5060;tag=38508459db22dfd
To: 19057412945 sip:[email protected]:5060;tag=as3b16b3fd
Call-ID: [email protected]
CSeq: 1806834509 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4910acc2"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
Found user '1000’
pbx*CLI>
<— SIP read from 192.168.31.112:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.31.112;branch=z9hG4bKb1fd1d1b8
Max-Forwards: 70
Content-Length: 0
To: 19057412945 sip:[email protected]:5060;tag=as3b16b3fd
From: sip:[email protected]:5060;tag=38508459db22dfd
Call-ID: [email protected]
CSeq: 1806834509 ACK
User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

<------------->
— (9 headers 0 lines) —
pbx*CLI>
<— SIP read from 192.168.31.112:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.31.112;branch=z9hG4bKed6c796b0
Max-Forwards: 70
Content-Length: 565
To: 19057412945 sip:[email protected]:5060
From: sip:[email protected]:5060;tag=38508459db22dfd
Call-ID: [email protected]
CSeq: 1806834510 INVITE
Supported: timer
Allow-Events: talk,hold,conference
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
Contact: sip:[email protected];transport=udp
Content-Type: application/sdp
Supported: replaces
Proxy-Authorization:Digest response=“f200f29ed55d0a44c12242c8aaaf37c4”,username=“1000”,realm=“asterisk”,nonce=“4910acc2”,algorithm=MD5,uri="sip:[email protected]:5060"
User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

v=0
o=MxSIP 0 1049955853 IN IP4 192.168.31.112
s=SIP Call
c=IN IP4 192.168.31.112
t=0 0
m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 BV16/8000
a=rtpmap:102 BV32/16000
a=rtpmap:107 L16/16000
a=rtpmap:104 PCMU/16000
a=rtpmap:105 PCMA/16000
a=rtpmap:106 L16/8000
a=rtpmap:97 G726-16/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:99 G726-40/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=silenceSupp:on - - - -

<------------->
— (16 headers 23 lines) —
Sending to 192.168.31.112 : 5060 (NAT)
Using INVITE request as basis request - [email protected]
Found user '1000’
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 102
Found RTP audio format 107
Found RTP audio format 104
Found RTP audio format 105
Found RTP audio format 106
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 2
Found RTP audio format 99
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.31.112:3000
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found unknown media description format BV16 for ID 96
Found unknown media description format BV32 for ID 102
Found audio description format L16 for ID 107
Found audio description format PCMU for ID 104
Found audio description format PCMA for ID 105
Found audio description format L16 for ID 106
Found unknown media description format G726-16 for ID 97
Found unknown media description format G726-24 for ID 98
Found audio description format G726-32 for ID 2
Found unknown media description format G726-40 for ID 99
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x94c (ulaw|alaw|g726|slin|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.31.112:3000
Looking for 19057412945 in from-internal (domain 192.168.31.31)
list_route: hop: sip:[email protected];transport=udp

<— Transmitting (NAT) to 192.168.31.112:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.31.112;branch=z9hG4bKed6c796b0;received=192.168.31.112
From: sip:[email protected]:5060;tag=38508459db22dfd
To: 19057412945 sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 1806834510 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
– Executing [[email protected]:1] ResetCDR(“SIP/1000-09285640”, “”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/1000-09285640”, “”) in new stack
– Executing [[email protected]:3] Wait(“SIP/1000-09285640”, “1”) in new stack
– Executing [[email protected]:4] Playback(“SIP/1000-09285640”, “silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer”) in new stack
Audio is at 192.168.31.31 port 13330
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
pbx*CLI>
<— Transmitting (NAT) to 192.168.31.112:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.31.112;branch=z9hG4bKed6c796b0;received=192.168.31.112
From: sip:[email protected]:5060;tag=38508459db22dfd
To: 19057412945 sip:[email protected]:5060;tag=as067f8184
Call-ID: [email protected]
CSeq: 1806834510 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 2686 2686 IN IP4 192.168.31.31
s=session
c=IN IP4 192.168.31.31
t=0 0
m=audio 13330 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– <SIP/1000-09285640> Playing ‘silence/1’ (language ‘en’)
– <SIP/1000-09285640> Playing ‘cannot-complete-as-dialed’ (language ‘en’)
– <SIP/1000-09285640> Playing ‘check-number-dial-again’ (language ‘en’)
– Executing [[email protected]:5] Wait(“SIP/1000-09285640”, “1”) in new stack
– Executing [[email protected]:6] Congestion(“SIP/1000-09285640”, “20”) in new stack

<— Transmitting (NAT) to 192.168.31.112:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.31.112;branch=z9hG4bKed6c796b0;received=192.168.31.112
From: sip:[email protected]:5060;tag=38508459db22dfd
To: 19057412945 sip:[email protected]:5060;tag=as067f8184
Call-ID: [email protected]
CSeq: 1806834510 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0

<------------>
== Spawn extension (from-internal, 19057412945, 6) exited non-zero on ‘SIP/1000-09285640’
– Executing [[email protected]:1] Macro(“SIP/1000-09285640”, “hangupcall”) in new stack
– Executing [[email protected]:1] ResetCDR(“SIP/1000-09285640”, “vw”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/1000-09285640”, “”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/1000-09285640”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [[email protected]:6] GotoIf(“SIP/1000-09285640”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/1000-09285640”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [[email protected]:11] Hangup(“SIP/1000-09285640”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/1000-09285640’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-09285640’
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
pbx*CLI>
<— SIP read from 192.168.31.112:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.31.112;branch=z9hG4bKed6c796b0
Max-Forwards: 70
Content-Length: 0
To: 19057412945 sip:[email protected]:5060;tag=as067f8184
From: sip:[email protected]:5060;tag=38508459db22dfd
Call-ID: [email protected]
CSeq: 1806834510 ACK
Proxy-Authorization:Digest response=“f200f29ed55d0a44c12242c8aaaf37c4”,username=“1000”,realm=“asterisk”,nonce=“4910acc2”,algorithm=MD5,uri="sip:[email protected]:5060"
User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

Thanks for pointing me in the right direction.

I decided to play with the dial patterns after reading extensively about them.

It turns out the dial pattern I was using was incorrect. I changed it from:
1|nxxxxxx to 1nxxnxxxxxx.

It was just like magic and I was making outbound calls again.

Now its time to test the inbound routes.

Thanks again.