Outbound Calls not working

Good Morning… Having trouble with outbound calls, In-bound working good… doesn’t look like it picking outbound route, have been scouring the forums and learning to read the debug logs, just not seeing it yet…

Thanks in advance

Sip_General
accept_outofcall_messages=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
faxdetect=yes
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.7.0)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
callevents=no
jbenable=no
defaultexpiry=120
allowguest=yes
srvlookup=no
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
notifyhold=yes
rtpkeepalive=120
g726nonstandard=no
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=600
notifyringing=yes
checkmwi=10
nat=yes
externip=64.60.155.changed
localnet=10.0.0.0/255.0.0.0

Extensions
[1500]
deny=0.0.0.0/0.0.0.0
secret=changed
dtmfmode=auto
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=peer
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=tcp,udp,tls
avpf=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/1500
[email protected]
permit=0.0.0.0/0.0.0.0
callerid=Test 1500 <1500>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

Trunk
[lvl3in]
host=4.55.18.225
type=peer
context=from-trunk

[lvl3out]
host=4.55.18.225
type=peer
context=from-trunk
insecure=port,invite

Sip Debug

localhost*CLI> sip set debug on
SIP Debugging enabled
Reliably Transmitting (no NAT) to 10.0.0.13:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK7370f9bb
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as3fbe2a7c
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.7.0)
Date: Mon, 03 Feb 2014 19:15:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:10.0.0.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bK7370f9bb
From: “Unknown” sip:[email protected];tag=as3fbe2a7c
To: sip:[email protected]:5060;tag=1542523970
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1450 1.0.5.23
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:10.0.0.13:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.13:5060;branch=z9hG4bK830886150;rport
Route: sip:10.0.0.5:5060;lr
From: sip:[email protected];tag=1116999234
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 90 INVITE
Contact: sip:[email protected]:5060
Max-Forwards: 70
User-Agent: Grandstream GXP1450 1.0.5.23
Privacy: none
P-Preferred-Identity: sip:[email protected]
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 396

v=0
o=1500 8000 8000 IN IP4 10.0.0.13
s=SIP Call
c=IN IP4 10.0.0.13
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (17 headers 19 lines) —
Sending to 10.0.0.13:5060 (NAT)
Sending to 10.0.0.13:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘1500’ for ‘1500’ from 10.0.0.13:5060

<— Reliably Transmitting (no NAT) to 10.0.0.13:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.13:5060;branch=z9hG4bK830886150;received=10.0.0.13;rport=5060
From: sip:[email protected];tag=1116999234
To: sip:[email protected];tag=as15420c2d
Call-ID: [email protected]
CSeq: 90 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="23f15f55"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:10.0.0.13:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.13:5060;branch=z9hG4bK830886150;rport
Route: sip:10.0.0.5:5060;lr
From: sip:[email protected];tag=1116999234
To: sip:[email protected];tag=as15420c2d
Call-ID: [email protected]
CSeq: 90 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.0.0.13:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.13:5060;branch=z9hG4bK1750238699;rport
Route: sip:10.0.0.5:5060;lr
From: sip:[email protected];tag=1116999234
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 91 INVITE
Contact: sip:[email protected]:5060
Authorization: Digest username=“1500”, realm=“asterisk”, nonce=“23f15f55”, uri="sip:[email protected]", response=“faa3b9dcc4f53d9d0f562c8b2ad7cfb4”, algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXP1450 1.0.5.23
Privacy: none
P-Preferred-Identity: sip:[email protected]
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 396

v=0
o=1500 8000 8000 IN IP4 10.0.0.13
s=SIP Call
c=IN IP4 10.0.0.13
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (18 headers 19 lines) —
Sending to 10.0.0.13:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘1500’ for ‘1500’ from 10.0.0.13:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 97
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.0.13:5004
Looking for 95878919 in from-internal (domain 10.0.0.5)
list_route: hop: sip:[email protected]:5060

<— Transmitting (no NAT) to 10.0.0.13:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.13:5060;branch=z9hG4bK1750238699;received=10.0.0.13;rport=5060
From: sip:[email protected];tag=1116999234
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 91 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
– Executing [[email protected]:1] ResetCDR(“SIP/1500-00000038”, “”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/1500-00000038”, “”) in new stack
– Executing [[email protected]:3] Progress(“SIP/1500-00000038”, “”) in new stack
Audio is at 17404
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 10.0.0.13:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.13:5060;branch=z9hG4bK1750238699;received=10.0.0.13;rport=5060
From: sip:[email protected];tag=1116999234
To: sip:[email protected];tag=as2d5af16e
Call-ID: [email protected]
CSeq: 91 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 251

v=0
o=root 1392516839 1392516839 IN IP4 10.0.0.5
s=Asterisk PBX 11.7.0
c=IN IP4 10.0.0.5
t=0 0
m=audio 17404 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=ptime:20
a=sendrecv

<------------>
– Executing [[email protected]:4] Wait(“SIP/1500-00000038”, “1”) in new stack
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘CHAN_START’,{ts ‘2014-02-03 11:15:09’},‘Test 1500’,‘1500’,’’,’’,’’,‘95878919’,‘from-internal’,‘SIP/1500-00000038’,’’,’’,3,’’,‘1391454909.56’,‘1391454909.56’,’’,’’,’’)]
> 0x7f6c0404e450 – Probation passed - setting RTP source address to 10.0.0.13:5004
– Executing [[email protected]:5] Progress(“SIP/1500-00000038”, “”) in new stack
– Executing [[email protected]:6] Playback(“SIP/1500-00000038”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <SIP/1500-00000038> Playing ‘silence/1.ulaw’ (language ‘en’)
– <SIP/1500-00000038> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)

<— SIP read from UDP:10.0.0.13:5060 —>
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.13:5060;branch=z9hG4bK1750238699;rport
Route: sip:10.0.0.5:5060;lr
From: sip:[email protected];tag=1116999234
To: sip:[email protected]
Call-ID: 2012803[email protected]
CSeq: 91 CANCEL
Max-Forwards: 70
User-Agent: Grandstream GXP1450 1.0.5.23
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 10.0.0.13:5060 (no NAT)

<— Reliably Transmitting (no NAT) to 10.0.0.13:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.0.13:5060;branch=z9hG4bK1750238699;received=10.0.0.13;rport=5060
From: sip:[email protected];tag=1116999234
To: sip:[email protected];tag=as2d5af16e
Call-ID: [email protected]
CSeq: 91 INVITE
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<— Transmitting (no NAT) to 10.0.0.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.13:5060;branch=z9hG4bK1750238699;received=10.0.0.13;rport=5060
From: sip:[email protected];tag=1116999234
To: sip:[email protected];tag=as2d5af16e
Call-ID: [email protected]
CSeq: 91 CANCEL
Server: FPBX-2.11.0(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (from-internal, 95878919, 6) exited non-zero on ‘SIP/1500-00000038’
– Executing [[email protected]:1] Hangup(“SIP/1500-00000038”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1500-00000038’
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘HANGUP’,{ts ‘2014-02-03 11:15:13’},‘Test 1500’,‘1500’,‘1500’,’’,‘95878919’,‘h’,‘from-internal’,‘SIP/1500-00000038’,’’,’’,3,’’,‘1391454909.56’,‘1391454909.56’,’’,’’,’’)]
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘CHAN_END’,{ts ‘2014-02-03 11:15:13’},‘Test 1500’,‘1500’,‘1500’,’’,‘95878919’,‘h’,‘from-internal’,‘SIP/1500-00000038’,’’,’’,3,’’,‘1391454909.56’,‘1391454909.56’,’’,’’,’’)]
> [INSERT INTO cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES (‘LINKEDID_END’,{ts ‘2014-02-03 11:15:13’},‘Test 1500’,‘1500’,‘1500’,’’,‘95878919’,‘h’,‘from-internal’,‘SIP/1500-00000038’,’’,’’,3,’’,‘1391454909.56’,‘1391454909.56’,’’,’’,’’)]

<— SIP read from UDP:10.0.0.13:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.13:5060;branch=z9hG4bK1750238699;rport
Route: sip:10.0.0.5:5060;lr
From: sip:[email protected];tag=1116999234
To: sip:[email protected];tag=as2d5af16e
Call-ID: [email protected]
CSeq: 91 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: ACK
localhostCLI> sip set debug off
SIP Debugging Disabled
localhost
CLI>

also have you verified that your trunk definitions work for you sip provider?

Route Name level3
dial pattern 9|
trunk seq… level3

tried to make these test as simple as possible without options…

Level 3 tells me they only require the host

[lvl3out]
host=4.55.18.225
type=peer
context=from-trunk
insecure=port,invite

I have tried many different options including only host=
same result as about log file…

I did notice the “TO” section it is showing the internal ip address, should it not be showing the external ip that is setup on the sip settings page?. it seems to me the system would believe that is an internal call?