[SOLVED] Call cannot completed in outbound SIP trunk - Amazon Cloud Ubuntu 16.04 LTS

Asterisk Asterisk Support
1

Call cannot completed return BUSY to external call using a VOIP wholesale (checkbox.cc)

The caller number is 5000 and trying to call to my house number 551934518014 (number of Brazil)

sip.conf*

[general]
allowguest=no
context=basico
port=5060
bindport=5060
bindaddr=0.0.0.0
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
maxexpirey=360
defaultexpirey=120
disallow=all
allow=ulaw
externaddr=35.177.209.216
media_address=35.177.209.216
localnet=192.168.0.0/255.255.0.0
nat=force_rport,comedia
qualify=yes
register=>lmoraes.oliveira1219-GW:******@199.59.99.91

[checkbox]
type=peer
username=lmoraes.oliveira1219-GW
secret=******
domain=199.59.99.91
fromuser=lmoraes.oliveira1219-GW
fromdomain=199.59.99.91
host=199.59.99.91
qualify=yes
port=5060
nat=force_rport,comedia
disallow=all
allow=ulaw
dtmfmode=rfc2833
context=recebe_checkbox
canreinvite=yes

[5000]
type=friend
secret=*****
qualify=yes
nat=force_rport,comedia
host=dynamic
disallow=all
allow=ulaw
context=rotadesaida
canreinvite=yes

extensions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
IAXINFO=guest                                   ; IAXtel username/password
TRUNK=DAHDI/G2                                  ; Trunk interface
TRUNKMSD=1                                      ; MSD digits to strip (usually 1 or 0)

[rotadesaida]
exten => _X.,1,Dial(SIP/${EXTEN}@checkbox,50)
exten => _X.,2,congestion()
exten => _X.,102,busy()

[recebe_checkbox]
exten => checkbox,1,Goto(rotadeentrada,s,1)

[rotadeentrada]
exten => s,1,Dial(SIP/${EXTEN},50)

Complete log of SIP and RTM Debug ON

<--- SIP read from UDP:177.129.14.34:27436 --->
INVITE sip:551943518014@solaristelecom.ddns.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 177.129.14.34:27436;branch=z9hG4bK-524287-1---a4442f1fedd77411;rport
Max-Forwards: 70
Contact: <sip:5000@177.129.14.34:27436;transport=UDP>
To: <sip:551943518014@solaristelecom.ddns.net;transport=UDP>
From: <sip:5000@solaristelecom.ddns.net;transport=UDP>;tag=06eab136
Call-ID: IkgQfNSXNJhh2xlnpR8RXA..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rd82a609
Allow-Events: presence, kpml, talk
Content-Length: 243

v=0
o=Zoiper 0 0 IN IP4 177.129.14.34
s=Zoiper
c=IN IP4 177.129.14.34
t=0 0
m=audio 27466 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
Sending to 177.129.14.34:27436 (NAT)
Sending to 177.129.14.34:27436 (NAT)
Using INVITE request as basis request - IkgQfNSXNJhh2xlnpR8RXA..
Found peer '5000' for '5000' from 177.129.14.34:27436

<--- Reliably Transmitting (NAT) to 177.129.14.34:27436 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 177.129.14.34:27436;branch=z9hG4bK-524287-1---a4442f1fedd77411;received=177.129.14.34;rport=27436
From: <sip:5000@solaristelecom.ddns.net;transport=UDP>;tag=06eab136
To: <sip:551943518014@solaristelecom.ddns.net;transport=UDP>;tag=as2caa5a6a
Call-ID: IkgQfNSXNJhh2xlnpR8RXA..
CSeq: 1 INVITE
Server: Asterisk PBX 13.18.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30eeff3e"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'IkgQfNSXNJhh2xlnpR8RXA..' in 83264 ms (Method: INVITE)

<--- SIP read from UDP:177.129.14.34:27436 --->
ACK sip:551943518014@solaristelecom.ddns.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 177.129.14.34:27436;branch=z9hG4bK-524287-1---a4442f1fedd77411;rport
Max-Forwards: 70
To: <sip:551943518014@solaristelecom.ddns.net;transport=UDP>;tag=as2caa5a6a
From: <sip:5000@solaristelecom.ddns.net;transport=UDP>;tag=06eab136
Call-ID: IkgQfNSXNJhh2xlnpR8RXA..
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:177.129.14.34:27436 --->
INVITE sip:551943518014@solaristelecom.ddns.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 177.129.14.34:27436;branch=z9hG4bK-524287-1---e990ad657519ad3b;rport
Max-Forwards: 70
Contact: <sip:5000@177.129.14.34:27436;transport=UDP>
To: <sip:551943518014@solaristelecom.ddns.net;transport=UDP>
From: <sip:5000@solaristelecom.ddns.net;transport=UDP>;tag=06eab136
Call-ID: IkgQfNSXNJhh2xlnpR8RXA..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rd82a609
Authorization: Digest username="5000",realm="asterisk",nonce="30eeff3e",uri="sip:551943518014@solaristelecom.ddns.net;transport=UDP",response="7e8115c561dd54a516f60cad749cb029",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 243

v=0
o=Zoiper 0 0 IN IP4 177.129.14.34
s=Zoiper
c=IN IP4 177.129.14.34
t=0 0
m=audio 27466 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 177.129.14.34:27436 (NAT)
Using INVITE request as basis request - IkgQfNSXNJhh2xlnpR8RXA..
Found peer '5000' for '5000' from 177.129.14.34:27436
  == Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 177.129.14.34:27466
Looking for 551943518014 in rotadesaida (domain solaristelecom.ddns.net)
sip_route_dump: route/path hop: <sip:5000@177.129.14.34:27436;transport=UDP>

<--- Transmitting (NAT) to 177.129.14.34:27436 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 177.129.14.34:27436;branch=z9hG4bK-524287-1---e990ad657519ad3b;received=177.129.14.34;rport=27436
From: <sip:5000@solaristelecom.ddns.net;transport=UDP>;tag=06eab136
To: <sip:551943518014@solaristelecom.ddns.net;transport=UDP>
Call-ID: IkgQfNSXNJhh2xlnpR8RXA..
CSeq: 2 INVITE
Server: Asterisk PBX 13.18.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:551943518014@35.177.209.216:5060>
Content-Length: 0


<------------>
    -- Executing [551943518014@rotadesaida:1] Dial("SIP/5000-00000052", "SIP/551943518014@checkbox,50") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/551943518014@checkbox
    -- SIP/checkbox-00000053 redirecting info has changed, passing it to SIP/5000-00000052

<--- Transmitting (NAT) to 177.129.14.34:27436 --->
SIP/2.0 181 Call is being forwarded
Via: SIP/2.0/UDP 177.129.14.34:27436;branch=z9hG4bK-524287-1---e990ad657519ad3b;received=177.129.14.34;rport=27436
From: <sip:5000@solaristelecom.ddns.net;transport=UDP>;tag=06eab136
To: <sip:551943518014@solaristelecom.ddns.net;transport=UDP>;tag=as3b1d5cf9
Call-ID: IkgQfNSXNJhh2xlnpR8RXA..
CSeq: 2 INVITE
Server: Asterisk PBX 13.18.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:551943518014@35.177.209.216:5060>
Content-Length: 0


<------------>
    -- SIP/checkbox-00000053 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [551943518014@rotadesaida:2] Congestion("SIP/5000-00000052", "") in new stack

<--- Reliably Transmitting (NAT) to 177.129.14.34:27436 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 177.129.14.34:27436;branch=z9hG4bK-524287-1---e990ad657519ad3b;received=177.129.14.34;rport=27436
From: <sip:5000@solaristelecom.ddns.net;transport=UDP>;tag=06eab136
To: <sip:551943518014@solaristelecom.ddns.net;transport=UDP>;tag=as3b1d5cf9
Call-ID: IkgQfNSXNJhh2xlnpR8RXA..
CSeq: 2 INVITE
Server: Asterisk PBX 13.18.5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0


<------------>
  == Spawn extension (rotadesaida, 551943518014, 2) exited non-zero on 'SIP/5000-00000052'
[Jan  8 22:45:21] ERROR[1540]: cdr_csv.c:315 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied

<--- SIP read from UDP:177.129.14.34:27436 --->
ACK sip:551943518014@solaristelecom.ddns.net;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 177.129.14.34:27436;branch=z9hG4bK-524287-1---e990ad657519ad3b;rport
Max-Forwards: 70
To: <sip:551943518014@solaristelecom.ddns.net;transport=UDP>;tag=as3b1d5cf9
From: <sip:5000@solaristelecom.ddns.net;transport=UDP>;tag=06eab136
Call-ID: IkgQfNSXNJhh2xlnpR8RXA..
CSeq: 2 ACK
Content-Length: 0
2

This is a relevant pice of log to find out what your issue is

3

The return of the command sip show registry is:

ip-172-31-3-42*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time
199.59.99.91:5060                       N      lmoraes.oliv       120 No Authentication
1 SIP registrations.

This means that the service is unavailable?

4

Registration it is when you dont have an static IP and is the way how you let you carrier know your current IP and how he can contact you, this is used for inbound calls for outbound calling you dont need registration

5

oh ok, thank you ambiorxg12

6

have you any idea, how I can search about this issue?

7

logs shows calls 181 Call is Being Forwarded and after that no answer , I suggest you contact your carrier

8

thank you ambiorixg12