Dialplan outgoing call

Hello, I am implanting the asterisk in my company, for a while almost all is working well. But I’m having problem with outgoing calls.
In this moment inbound call are ok via SIP Trunk, extensions are ok, but the outgoing calls don’t work.
I will show the part of the configuration files that interest.

hosts

127.0.0.1 VESfone

*this config I add becouse before the asterisk won’t get the name “VESfone”, and in my searches found this solution, but don’t have sure if is the correct.

sip.conf

[general]
bindaddr=0.0.0.0
bindport=5060
context=vesfone
disallow=all
allow=gsm
allow=alaw
dtmfmode=inband
register => 3832161260:4157649228@10.45.0.11/3832161260
[VESfone]
type=user
defaultuser=3832161260
secret=4157649228
disallow=all
allow=alaw
dtmfmode=inband
host=10.45.0.11
defaultip=10.45.0.11
context=fromPstn
insecure=invite,port
transport=udp
directmedia=no

extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=yes
priorityjumping=yes

[globals]
SAMUEL = SIP/2000
JP = SIP/2001
LUCAS = SIP/2002
SUPORTE03 = SIP/2003
MATHEUS = SIP/24
JOSSELE = SIP/2005
DOUGLAS = SIP/2006
VES = SIP/VESfone

[externas]
exten => _9.,1,Dial(${VES}/38${EXTEN})
exten => _9.,n,Playback(all-outgoing-lines-unavailable)
exten => _9.,n,Hangup()

asterisk -rvvvvv output while trying of outgoing call

Connected to Asterisk 18.13.0 currently running on ipbx (pid = 792)
       > Saved useragent "Telephone_TIP120" for peer 2005
       > Saved useragent "" for peer 2005
  == Using SIP RTP CoS mark 5
       > 0x7f3b6c065f70 -- Strict RTP learning after remote address set to: 192.168.1.10:40024
    -- Executing [991298847@all:1] Dial("SIP/2000-0000008a", "SIP/VESfone/38991298847") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/VESfone/38991298847
[Aug  3 13:03:22] NOTICE[913][C-0000007d]: chan_sip.c:24411 handle_response_invite: Failed to authenticate on INVITE to '"Gerente_Suporte_Samuel" <sip:2000@127.0.0.1>;tag=as79a8dd09'
    -- SIP/VESfone-0000008b is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [991298847@all:2] Playback("SIP/2000-0000008a", "all-outgoing-lines-unavailable") in new stack
       > 0x7f3b6c065f70 -- Strict RTP qualifying stream type: audio
       > 0x7f3b6c065f70 -- Strict RTP switching source address to 10.0.19.25:40024
    -- <SIP/2000-0000008a> Playing 'all-outgoing-lines-unavailable.gsm' (language 'en')
    -- Executing [991298847@all:3] Hangup("SIP/2000-0000008a", "") in new stack
  == Spawn extension (all, 991298847, 3) exited non-zero on 'SIP/2000-0000008a'

debug on output while trying of outgoing call

SIP Debugging enabled

<--- SIP read from UDP:10.0.19.25:64702 --->
INVITE sip:991298847@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-772dac597003af3c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef>
To: <sip:991298847@127.0.0.1:5060>
From: <sip:2000@127.0.0.1:5060>;tag=b7720517
Call-ID: YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 401

v=0
o=3cxVCE 357153195 89550630 IN IP4 192.168.1.10
s=3cxVCE Audio Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 40030 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=sendrecv
m=video 40028 RTP/AVP 34
c=IN IP4 10.0.19.25
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
Sending to 10.0.19.25:64702 (no NAT)
Sending to 10.0.19.25:64702 (no NAT)
Using INVITE request as basis request - YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
Found peer '2000' for '2000' from 10.0.19.25:64702

<--- Reliably Transmitting (no NAT) to 10.0.19.25:64702 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-772dac597003af3c-1---d8754z-;received=10.0.19.25;rport=64702
From: <sip:2000@127.0.0.1:5060>;tag=b7720517
To: <sip:991298847@127.0.0.1:5060>;tag=as20e43c3e
Call-ID: YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
CSeq: 1 INVITE
Server: Asterisk PBX 18.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7bf13bd7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.' in 7552 ms (Method: INVITE)

<--- SIP read from UDP:10.0.19.25:64702 --->
ACK sip:991298847@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-772dac597003af3c-1---d8754z-;rport
Max-Forwards: 70
To: <sip:991298847@127.0.0.1:5060>;tag=as20e43c3e
From: <sip:2000@127.0.0.1:5060>;tag=b7720517
Call-ID: YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.0.19.25:64702 --->
INVITE sip:991298847@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-ab22605588296710-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef>
To: <sip:991298847@127.0.0.1:5060>
From: <sip:2000@127.0.0.1:5060>;tag=b7720517
Call-ID: YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="2000",realm="asterisk",nonce="7bf13bd7",uri="sip:991298847@127.0.0.1:5060",response="b670298ac7a8e8bee6299868968560ba",algorithm=MD5
Content-Length: 401

v=0
o=3cxVCE 357153195 89550630 IN IP4 192.168.1.10
s=3cxVCE Audio Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 40030 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=sendrecv
m=video 40028 RTP/AVP 34
c=IN IP4 10.0.19.25
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 10.0.19.25:64702 (no NAT)
Using INVITE request as basis request - YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
Found peer '2000' for '2000' from 10.0.19.25:64702
  == Using SIP RTP CoS mark 5
Got SDP version 89550630 and unique parts [3cxVCE 357153195 IN IP4 192.168.1.10]
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
Capabilities: us - (gsm|alaw), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7f3b6c065f70 -- Strict RTP learning after remote address set to: 192.168.1.10:40030
Peer audio RTP is at port 192.168.1.10:40030
Looking for 991298847 in all (domain 127.0.0.1)
sip_route_dump: route/path hop: <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef>

<--- Transmitting (no NAT) to 10.0.19.25:64702 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-ab22605588296710-1---d8754z-;received=10.0.19.25;rport=64702
From: <sip:2000@127.0.0.1:5060>;tag=b7720517
To: <sip:991298847@127.0.0.1:5060>
Call-ID: YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
CSeq: 2 INVITE
Server: Asterisk PBX 18.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:991298847@127.0.0.1:5060>
Content-Length: 0


<------------>
    -- Executing [991298847@all:1] Dial("SIP/2000-0000008c", "SIP/VESfone/38991298847") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 17416
Adding codec gsm to SDP
Reliably Transmitting (no NAT) to 127.0.0.1:5060:
INVITE sip:38991298847@VESfone SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK426101eb
Max-Forwards: 70
From: "Gerente_Suporte_Samuel" <sip:2000@127.0.0.1>;tag=as42fc006f
To: <sip:38991298847@VESfone>
Contact: <sip:2000@127.0.0.1:5060>
Call-ID: 6bddbce01754c11f624f02c85ff72407@127.0.0.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.13.0
Date: Wed, 03 Aug 2022 13:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 189

v=0
o=root 1380872410 1380872410 IN IP4 127.0.0.1
s=Asterisk PBX 18.13.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 17416 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:300
a=sendrecv

---

<--- SIP read from UDP:127.0.0.1:5060 --->
INVITE sip:38991298847@VESfone SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK426101eb
Max-Forwards: 70
From: "Gerente_Suporte_Samuel" <sip:2000@127.0.0.1>;tag=as42fc006f
To: <sip:38991298847@VESfone>
Contact: <sip:2000@127.0.0.1:5060>
Call-ID: 6bddbce01754c11f624f02c85ff72407@127.0.0.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.13.0
Date: Wed, 03 Aug 2022 13:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 189

v=0
o=root 1380872410 1380872410 IN IP4 127.0.0.1
s=Asterisk PBX 18.13.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 17416 RTP/AVP 3
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:300
a=sendrecv
<------------->
--- (14 headers 10 lines) ---
Sending to 127.0.0.1:5060 (no NAT)
Sending to 127.0.0.1:5060 (no NAT)
Using INVITE request as basis request - 6bddbce01754c11f624f02c85ff72407@127.0.0.1:5060
Found peer '2000' for '2000' from 127.0.0.1:5060

<--- Reliably Transmitting (no NAT) to 127.0.0.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK426101eb;received=127.0.0.1
From: "Gerente_Suporte_Samuel" <sip:2000@127.0.0.1>;tag=as42fc006f
To: <sip:38991298847@VESfone>;tag=as45789815
Call-ID: 6bddbce01754c11f624f02c85ff72407@127.0.0.1:5060
CSeq: 102 INVITE
Server: Asterisk PBX 18.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="266e2e52"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '6bddbce01754c11f624f02c85ff72407@127.0.0.1:5060' in 7552 ms (Method: INVITE)

<--- SIP read from UDP:127.0.0.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK426101eb;received=127.0.0.1
From: "Gerente_Suporte_Samuel" <sip:2000@127.0.0.1>;tag=as42fc006f
To: <sip:38991298847@VESfone>;tag=as45789815
Call-ID: 6bddbce01754c11f624f02c85ff72407@127.0.0.1:5060
CSeq: 102 INVITE
Server: Asterisk PBX 18.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="266e2e52"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
    -- Called SIP/VESfone/38991298847
Transmitting (no NAT) to 127.0.0.1:5060:
ACK sip:38991298847@VESfone SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK426101eb
Max-Forwards: 70
From: "Gerente_Suporte_Samuel" <sip:2000@127.0.0.1>;tag=as42fc006f
To: <sip:38991298847@VESfone>;tag=as45789815
Contact: <sip:2000@127.0.0.1:5060>
Call-ID: 6bddbce01754c11f624f02c85ff72407@127.0.0.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.13.0
Content-Length: 0


---
[Aug  3 13:09:25] NOTICE[913][C-0000007f]: chan_sip.c:24411 handle_response_invite: Failed to authenticate on INVITE to '"Gerente_Suporte_Samuel" <sip:2000@127.0.0.1>;tag=as42fc006f'

<--- SIP read from UDP:127.0.0.1:5060 --->
ACK sip:38991298847@VESfone SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK426101eb
Max-Forwards: 70
From: "Gerente_Suporte_Samuel" <sip:2000@127.0.0.1>;tag=as42fc006f
To: <sip:38991298847@VESfone>;tag=as45789815
Contact: <sip:2000@127.0.0.1:5060>
Call-ID: 6bddbce01754c11f624f02c85ff72407@127.0.0.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.13.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- SIP/VESfone-0000008d is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [991298847@all:2] Playback("SIP/2000-0000008c", "all-outgoing-lines-unavailable") in new stack
Audio is at 18046
Adding codec gsm to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.0.19.25:64702 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-ab22605588296710-1---d8754z-;received=10.0.19.25;rport=64702
From: <sip:2000@127.0.0.1:5060>;tag=b7720517
To: <sip:991298847@127.0.0.1:5060>;tag=as36fd163d
Call-ID: YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
CSeq: 2 INVITE
Server: Asterisk PBX 18.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:991298847@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 293

v=0
o=root 966157132 966157132 IN IP4 127.0.0.1
s=Asterisk PBX 18.13.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 18046 RTP/AVP 3 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 0 RTP/AVP 34

<------------>
Really destroying SIP dialog '6bddbce01754c11f624f02c85ff72407@127.0.0.1:5060' Method: INVITE
       > 0x7f3b6c065f70 -- Strict RTP qualifying stream type: audio

<--- SIP read from UDP:10.0.19.25:64702 --->
ACK sip:991298847@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-a70f9d4db6429b08-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef>
To: <sip:991298847@127.0.0.1:5060>;tag=as36fd163d
From: <sip:2000@127.0.0.1:5060>;tag=b7720517
Call-ID: YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
CSeq: 2 ACK
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="2000",realm="asterisk",nonce="7bf13bd7",uri="sip:991298847@127.0.0.1:5060",response="b670298ac7a8e8bee6299868968560ba",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
       > 0x7f3b6c065f70 -- Strict RTP switching source address to 10.0.19.25:40030
    -- <SIP/2000-0000008c> Playing 'all-outgoing-lines-unavailable.gsm' (language 'en')

<--- SIP read from UDP:10.0.19.25:64702 --->
REGISTER sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-593eef3e8c20cc48-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef>
To: <sip:2000@127.0.0.1:5060>
From: <sip:2000@127.0.0.1:5060>;tag=45324b1c
Call-ID: MDM2OGU4ZTVmMWE2NmNiYzU3NmZmNWQ3MTcxNmU4Mzk.
CSeq: 131 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="2000",realm="asterisk",nonce="339564ab",uri="sip:127.0.0.1:5060",response="c59a6487a12f7930e51d8d711b7a29d2",algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.19.25:64702 (no NAT)
Sending to 10.0.19.25:64702 (no NAT)

<--- Transmitting (no NAT) to 10.0.19.25:64702 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-593eef3e8c20cc48-1---d8754z-;received=10.0.19.25;rport=64702
From: <sip:2000@127.0.0.1:5060>;tag=45324b1c
To: <sip:2000@127.0.0.1:5060>;tag=as22919eb4
Call-ID: MDM2OGU4ZTVmMWE2NmNiYzU3NmZmNWQ3MTcxNmU4Mzk.
CSeq: 131 REGISTER
Server: Asterisk PBX 18.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78dfd710"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'MDM2OGU4ZTVmMWE2NmNiYzU3NmZmNWQ3MTcxNmU4Mzk.' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.0.19.252:5060 --->

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

<--- SIP read from UDP:10.0.19.25:64702 --->
REGISTER sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-f90573401216a635-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef>
To: <sip:2000@127.0.0.1:5060>
From: <sip:2000@127.0.0.1:5060>;tag=45324b1c
Call-ID: MDM2OGU4ZTVmMWE2NmNiYzU3NmZmNWQ3MTcxNmU4Mzk.
CSeq: 132 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="2000",realm="asterisk",nonce="78dfd710",uri="sip:127.0.0.1:5060",response="133254d344d6435eb31fcb058c5d4105",algorithm=MD5
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.19.25:64702 (no NAT)
Reliably Transmitting (no NAT) to 10.0.19.25:64702:
OPTIONS sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3197370f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as2635ded1
To: <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef>
Contact: <sip:asterisk@127.0.0.1:5060>
Call-ID: 07172ae1653743ff2ccc551a41e397fc@127.0.0.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.13.0
Date: Wed, 03 Aug 2022 13:09:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 10.0.19.25:64702 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.19.25:64702;branch=z9hG4bK-d8754z-f90573401216a635-1---d8754z-;received=10.0.19.25;rport=64702
From: <sip:2000@127.0.0.1:5060>;tag=45324b1c
To: <sip:2000@127.0.0.1:5060>;tag=as22919eb4
Call-ID: MDM2OGU4ZTVmMWE2NmNiYzU3NmZmNWQ3MTcxNmU4Mzk.
CSeq: 132 REGISTER
Server: Asterisk PBX 18.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 120
Contact: <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef>;expires=120
Date: Wed, 03 Aug 2022 13:09:27 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'MDM2OGU4ZTVmMWE2NmNiYzU3NmZmNWQ3MTcxNmU4Mzk.' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.0.19.25:64702 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3197370f
Contact: <sip:10.0.19.25:64702>
To: <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef>;tag=675c1023
From: "asterisk"<sip:asterisk@127.0.0.1>;tag=as2635ded1
Call-ID: 07172ae1653743ff2ccc551a41e397fc@127.0.0.1:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
Allow-Events: presence, message-summary, tunnel-info
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '07172ae1653743ff2ccc551a41e397fc@127.0.0.1:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 10.0.19.7:5060:
OPTIONS sip:2006@10.0.19.7:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK18738eea
Max-Forwards: 70
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as72bfaadb
To: <sip:2006@10.0.19.7:5060>
Contact: <sip:asterisk@127.0.0.1:5060>
Call-ID: 359658c15034c1c6223c6f79043159ae@127.0.0.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 18.13.0
Date: Wed, 03 Aug 2022 13:09:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.19.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK18738eea
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as72bfaadb
To: <sip:2006@10.0.19.7:5060>;tag=48946000
Call-ID: 359658c15034c1c6223c6f79043159ae@127.0.0.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1405 1.0.7.4
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '359658c15034c1c6223c6f79043159ae@127.0.0.1:5060' Method: OPTIONS
    -- Executing [991298847@all:3] Hangup("SIP/2000-0000008c", "") in new stack
  == Spawn extension (all, 991298847, 3) exited non-zero on 'SIP/2000-0000008c'
Scheduling destruction of SIP dialog 'YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.' in 7552 ms (Method: ACK)
set_destination: Parsing <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef> for address/port to send to
set_destination: set destination to 10.0.19.25:64702
Reliably Transmitting (no NAT) to 10.0.19.25:64702:
BYE sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0f46f263;rport
Max-Forwards: 70
From: <sip:991298847@127.0.0.1:5060>;tag=as36fd163d
To: <sip:2000@127.0.0.1:5060>;tag=b7720517
Call-ID: YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
CSeq: 102 BYE
User-Agent: Asterisk PBX 18.13.0
Proxy-Authorization: Digest username="2000", realm="asterisk", algorithm=MD5, uri="sip:127.0.0.1", nonce="7bf13bd7", response="08e1b6d8644145d54c8ddc32d155495b"
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


---

<--- SIP read from UDP:10.0.19.25:64702 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK0f46f263;rport=5060
Contact: <sip:2000@10.0.19.25:64702;rinstance=5442d493d4b559ef>
To: <sip:2000@127.0.0.1:5060>;tag=b7720517
From: <sip:991298847@127.0.0.1:5060>;tag=as36fd163d
Call-ID: YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.
CSeq: 102 BYE
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'YmYwZDQ2OWJkODNkNzYwOTQ3NjJhMTUyOTViNjA1YzE.' Method: ACK

Reminiscing that the trunk is registered correctly and inboud calls are 100% ok.

I suppose the have some problem in the configuration, but I don’t found.
I will be very thankful if somebody can help me.

I find this strange, because you are getting congestion, rather than unavailable, without apparently sending any request to the correct IP.

Please note that you should be planning to move to chan_pjsip, as chan_sip is deprecated, only community supported, and in practice unsupported, and it is planned to completely remove it in next year’s release.

dtmfmode=inband is unusual and is incompatible with gsm (general section) as an allowed codec, especially as the first choice codec.

defaultip is only relevant for inbound registrations.

type=user creates a security problem when used with outbound registration, and, to be honest, I don’t understand why incoming calls are working, but I’d need to see an incoming call from the trunk, to be sure.

I don’t think that type=user can be used for outbound calls and so I think that Asterisk is falling back to treating VESfone as a simple domain name, but your hosts file has the wrong IP address for it. You should delete it from /etc/hosts.

I believe you should be using type=peer here. Actually you should normally use type=peer everywhere in sip.conf, except if you have multiple endpoints on the same IP address.

I’d therefore say this is not a dialplan problem.

Thanks friend, may be because the type really. Actually the type was peer, I that requested at operator change to user, because I was using Issabel and didn’t get register while type was peer.
But I gave up the Issabel and now I’m using pure asterisk.
I will request the operator to change to peer and I will try again, if continue with the problem, I’m back here.
Thanks very much.

type= isn’t something that the iITSP controls.

There may be cases where you need type=friend, which is basically a simultaneous type=peer and type=user, but they are relatively rare, and type=friend is very much overused in sip.conf.

Humm I don’t know where you are getting references for configuration, but it seems quite newbie.

First of all incoming are working because you set “insecure=invite,port” thus not requiring authentication on the incoming INVITE.

You set VESfone as host name resolution to ip address 127.0.0.1 where Asterisk is running therefore no way to make it work.

Have many things to correct and make clean, first start with SIP Trunk provider, your Sip provider must pass detailed information about your system configuration (host ip address, type peer, host or whatever) and set all on sip context profile.

Same for extension outgoing calls dialplan.

Something like;

…… Dial(SIP/${EXTEN}@trunk-provider

Good luck.

ITSPs generally don’t cooperate with inbound authentication, which is why inecure= so common. The port part is rarely correct, but the natural reaction of providers, when providing recommendations, is to turn off as much security as possible, and there used to be insecure=very. When this was removed, people just converted it to the equivalent without thinking.

The real problem here is how is Asterisk to match the incoming call against the section. For type=user, it would only match if the From user was VESfone, but it is much more likely to be the caller ID, I’m not sure if the default for allowguest was ever changed to no, but he may well be getting guest matches for incoming calls.

In my view, there has been a better option to insecure=invite, for a long time namely using remotesecret, rather than secret, but it didn’t exist when the original that has been copied and pasted by everyone was created.

Using user only makes sense if the ITSP has many source addresses and sets the account name in the From header.