Problem with outgoing calls

Hi! I am trying to get my Freepbx working. Created trunk for my Voip provider, inbound and outbound routes, extension for my softphone. In a freepbx status and log files i see that my trunk registers, i also can see that my softphone registered too. But when i make a call i get an error Server unreachable. I asked my provider for logs, but he says that he see only registration and doesnt see any calls. So i think that the problem is in outbound routes. I guess that the problem can also be connected with user context. As you see in trunk i choose my sip id from provider, but should i create some context in my freepbx, some dial plan or something or if i didnt specify any dial plans all calls would go by the outbound routes (to my providers trunk)? Maybe i should make a dial plan in my outbound routes so the server knows that if i call number from dial plan than give it to my providers trunk? What dialplan should i make if i need to call to Ukraine 38 code 050 mobil e operator 2030822 number. Should it be - prepend 38 prefix XNX and number XXXXXXX ?

Please make some advice of what is wrong in my configuration. Thanks for any help and advices. Good day to everyone and thanks for attention.

So, configuration is next:

Trunk
Trunk Name sipmarket
PEER Details
type=friend
secret=my account password
username=137739
host=sipmarket.net
insecure=invite
fromdomain=sipmarket.net
fromuser=137739
disallow=all
allow=alaw&ulaw&g729
canreinvite=no
dtmfmode=rfc2833

USER Context 137739
type=friend
secret=pass
username=137739
host=sipmarket.net
insecure=invite
fromdomain=sipmarket.net
fromuser=137739
disallow=all
allow=alaw&ulaw&g729
canreinvite=no
dtmfmode=rfc2833

Register String
137739:[email protected]/137739

Outbound Route

Route Name sipmarket
Trunk Sequence for Matched Routes? Have chosen sipmarket trunk

Inbound route

Has chosen my extension

Here is the debug info, when i make a call from my softphone.

<— SIP read from UDP:91.214.177.120:51097 —>

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

<— SIP read from UDP:91.214.177.120:51097 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 95.215.44.108:5060;branch=z9hG4bK189b9e9b
Contact: sip:10.202.82.59:60487
To: sip:[email protected]:51097;transport=UDP;rinstance=95d01b3db3b4d5da;tag=
6777bd4e
From: "Unknown"sip:[email protected];tag=as5c87611c
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, IN
FO, MESSAGE
Supported: replaces
Allow-Events: presence, message-summary, tunnel-info
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:506
0’ Method: OPTIONS

<— SIP read from UDP:91.214.177.120:51097 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.82.59:60487;branch=z9hG4bK-d8754z-b2078b0e3255db69-1—d
8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:63315;transport=UDP
To: sip:[email protected]:5060
From: sip:[email protected]:5060;tag=e650d868
Call-ID: ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVmN2Q.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, IN
FO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 424

v=0
o=3cxVCE 168694425 147670890 IN IP4 91.214.177.120
s=3cxVCE Audio Call
c=IN IP4 91.214.177.120
t=0 0
m=audio 58515 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=rtcp:64142
a=sendrecv
m=video 50032 RTP/AVP 34
c=IN IP4 91.214.177.120
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv
<------------->
— (13 headers 19 lines) —
Sending to 91.214.177.120:51097 (NAT)
Sending to 91.214.177.120:51097 (NAT)
Using INVITE request as basis request - ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVm
N2Q.
Found peer ‘300’ for ‘300’ from 91.214.177.120:51097

<— Reliably Transmitting (no NAT) to 91.214.177.120:51097 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.202.82.59:60487;branch=z9hG4bK-d8754z-b2078b0e3255db69-1—d
8754z-;received=91.214.177.120;rport=51097
From: sip:[email protected]:5060;tag=e650d868
To: sip:[email protected]:5060;tag=as002b67e5
Call-ID: ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVmN2Q.
CSeq: 1 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7b6394f2"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVmN2
Q.’ in 11456 ms (Method: INVITE)

<— SIP read from UDP:91.214.177.120:51097 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.82.59:60487;branch=z9hG4bK-d8754z-b2078b0e3255db69-1—d
8754z-;rport
Max-Forwards: 70
To: sip:[email protected]:5060;tag=as002b67e5
From: sip:[email protected]:5060;tag=e650d868
Call-ID: ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVmN2Q.
CSeq: 1 ACK
Content-Length: 0

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

<— SIP read from UDP:91.214.177.120:51097 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.82.59:60487;branch=z9hG4bK-d8754z-1b25a42d6642c00b-1—d
8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:63315;transport=UDP
To: sip:[email protected]:5060
From: sip:[email protected]:5060;tag=e650d868
Call-ID: ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVmN2Q.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, IN
FO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username=“300”,realm=“asterisk”,nonce=“7b6394f2”,uri=“sip:
[email protected]:5060”,response=“2c543ead90a33cf9b7dc12756bc94b7c”,alg
orithm=MD5
Content-Length: 424

v=0
o=3cxVCE 168694425 147670890 IN IP4 91.214.177.120
s=3cxVCE Audio Call
c=IN IP4 91.214.177.120
t=0 0
m=audio 58515 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=rtcp:64142
a=sendrecv
m=video 50032 RTP/AVP 34
c=IN IP4 91.214.177.120
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv
<------------->
— (14 headers 19 lines) —
Sending to 91.214.177.120:51097 (no NAT)
Using INVITE request as basis request - ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVm
N2Q.
Found peer ‘300’ for ‘300’ from 91.214.177.120:51097
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
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 GSM for ID 3
Found audio description format telephone-event for ID 101
Found RTP video format 34
Found video description format H263 for ID 34
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw)/video=(h263)/text=(
nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon
e-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 91.214.177.120:58515
Looking for 380502030822 in from-internal (domain 95.215.44.108)
list_route: hop: sip:[email protected]:63315;transport=UDP

<— Transmitting (no NAT) to 91.214.177.120:51097 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.202.82.59:60487;branch=z9hG4bK-d8754z-1b25a42d6642c00b-1—d
8754z-;received=91.214.177.120;rport=51097
From: sip:[email protected]:5060;tag=e650d868
To: sip:[email protected]:5060
Call-ID: ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVmN2Q.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
Audio is at 17502
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 91.214.177.120:51097 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.202.82.59:60487;branch=z9hG4bK-d8754z-1b25a42d6642c00b-1—d
8754z-;received=91.214.177.120;rport=51097
From: sip:[email protected]:5060;tag=e650d868
To: sip:[email protected]:5060;tag=as34fbb1a8
Call-ID: ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVmN2Q.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 1275610450 1275610450 IN IP4 95.215.44.108
s=Asterisk PBX 11.11.0
c=IN IP4 95.215.44.108
t=0 0
m=audio 17502 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
m=video 0 RTP/AVP 34

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

<— Reliably Transmitting (no NAT) to 91.214.177.120:51097 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.202.82.59:60487;branch=z9hG4bK-d8754z-1b25a42d6642c00b-1—d
8754z-;received=91.214.177.120;rport=51097
From: sip:[email protected]:5060;tag=e650d868
To: sip:[email protected]:5060;tag=as34fbb1a8
Call-ID: ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVmN2Q.
CSeq: 2 INVITE
Server: FPBX-2.11.0(11.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H, MESSAGE
Supported: replaces, timer
Content-Length: 0

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

<— SIP read from UDP:91.214.177.120:51097 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.202.82.59:60487;branch=z9hG4bK-d8754z-1b25a42d6642c00b-1—d
8754z-;rport
Max-Forwards: 70
To: sip:[email protected]:5060;tag=as34fbb1a8
From: sip:[email protected]:5060;tag=e650d868
Call-ID: ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVmN2Q.
CSeq: 2 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘ZTZkMTgyMmMwM2E5MGJlODQ1Y2RhZjdlODI1NTVmN2Q.’ Meth
od: ACK
25d26292c9*CLI>

How about a SIP trace when you try the call from Astersk.

Doesn’t the carrier have a sample Asterisk config?