Getting "Everyone is busy/congested at this time (1:0/0/1)"

Asterisk Asterisk SIP
1

Newbie here. I installed and configured asterisk server using sip trunking from voipvoip.com , but i cant seem to make outbound calls, I’m getting this busy/congested error. I tried looking up on forums and other sites for any possible solution but couldn’t resolve it.

Here are my asterisk configurations:

=========PJSIP.CONF=========

[global]
type=global

[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
external_media_address=xxx.xx.xx.xx
external_signaling_address=xxx.xxx.xxx.xxx
allow_reload=no
tos=cs3
cos=3
local_net=xx.xx.xx.xx/24


[VoIPVoIP-auth]
type = auth
auth_type = userpass
username = 5551231234
password = nottellingyou      ; your VoIPVoIP password

[VoIPVoIP-aor]
type = aor
contact = sip:sip3.voipvoip.com ; SIP trunk provider's URI

; Add any other AOR settings here if needed

[VoIPVoIP]
type = aor
contact = sip:sip3.voipvoip.com ; SIP trunk provider's URI

; Add any other AOR settings here if needed

[VoIPVoIP-endpoint]
type = endpoint
context = from-trunk
transport = 0.0.0.0-udp
dtmf_mode = rfc4733
disallow = all
allow = ulaw
allow = alaw
aors = VoIPVoIP-aor
auth = VoIPVoIP-auth
outbound_auth = VoIPVoIP-auth
outbound_proxy = sip:sip3.voipvoip.com  ; Replace with your SIP trunk provider's URI
from_domain = sip3.voipvoip.com         ; Replace with your SIP trunk provider's domain
from_user = 5551231234

[VoIPVoIP-registration]
type=registration
transport=0.0.0.0-udp ; Replace with your transport configuration
outbound_auth=VoIPVoIP-auth
server_uri=sip:sip3.voipvoip.com ; SIP trunk provider's URI
client_uri=sip:5551231234@sip3.voipvoip.com ; Your VoIPVoIP account URI
retry_interval=60 ; You can adjust this retry interval as needed
forbidden_retry_interval=300 ; You can adjust this retry interval as needed
fatal_retry_interval=3600 ; You can adjust this retry interval as needed


[VoIPVoIP-identify]
type=identify
endpoint=VoIPVoIP-endpoint
match=sip3.voipvoip.com



; Add any other endpoint settings here if needed


;--------------------------
;       ENDPOINT TEMPLATE
;--------------------------

[endpoint-basic](!)
type=endpoint
transport=0.0.0.0-udp
context=from-internal
disallow=all
allow=ulaw
allow=alaw
direct_media=no

[auth-userpass](!)
type=auth
auth_type=userpass
password=asimplepass

[aor-single-reg](!)
type=aor
max_contacts=1

;---------------------
;       EXTENSION 7001
;---------------------

[7001](endpoint-basic)
callerid= "7001" <7001>
auth= 7001
aors=7001

[7001](auth-userpass)
username=7001

[7001](aor-single-reg)
max_contacts=2


==============extensions.conf==========
[from-trunk]
exten => _+1NXXXXXXXXX,1,Dial(PJSIP/${EXTEN})

[from-internal]
exten => _NXXNXXXXXX,1,Set(CALLERID(all)="bob" <+15551231234>)
same => n,Dial(PJSIP/+1${EXTEN}@VoIPVoIP-endpoint)
same => n(end),Hangup()

Registration status VoIPVoIP Trunk : Registered
Endpoints are available and reachable.

On wireshark dump , i noticed there’s a 484 address incomplete but i dont know what to make of it or what’s causing it.

image
image852×263 60.2 KB

2

The provider sent back a SIP response of “484 Address Incomplete” meaning they didn’t like something about your call attempt, most likely the number dialled. The dialled number also doesn’t match your dialplan from what I can tell, because I think it starts with an “8” when your dialplan shows it should start with a “+”.

Oh, unless this is actually the call into Asterisk. You’ve redacted enough and haven’t given enough context that I assumed what you provided was actually on the call leg from Asterisk to the provider.

You’d really want to provide the Asterisk console output alongside the SIP trace from there (pjsip set logger on).

3

Hi @jcolp ,

when i dial a test number 8042221111 , the asterisk console output without logger on is -

 Executing [8042221111@from-internal:1] Set("PJSIP/7001-00000012", "CALLERID(all)="bob" <+15551231234>") in new stack
    -- Executing [8042221111@from-internal:2] Dial("PJSIP/7001-00000012", "PJSIP/+18042221111@VoIPVoIP-endpoint") in new stack
    -- Called PJSIP/+18042221111@VoIPVoIP-endpoint
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [8042221111@from-internal:3] Hangup("PJSIP/7001-00000012", "") in new stack

is this a dialplan issue?

4

You would need to show the SIP trace between Asterisk and the provider.

5

0/0/1 means 0 busy, 0 congested, 1 unavailable.

Unavailable could mean not registered, not responding to qualify, or invalid number, etc. You need to know the primary error to understand the actual failure.

6

@jcolp @david551

ip-172-31-41-232*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (1052 bytes) from UDP:123.253.126.205:44354 --->
INVITE sip:8144098500@13.200.0.116:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.69.139:44354;branch=z9hG4bK-524287-1---e25ffcb5ce799478;rport
Max-Forwards: 70
Contact: <sip:7001@123.253.126.205:44354;transport=UDP>
To: <sip:8144098500@13.200.0.116:5060>
From: <sip:7001@13.200.0.116:5060;transport=UDP>;tag=7b820379
Call-ID: vVWiZBwt0yMGTPU0nbFKWw..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 346

v=0
o=Z 0 21006532 IN IP4 123.253.126.205
s=Z
c=IN IP4 123.253.126.205
t=0 0
m=audio 32816 RTP/AVP 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP response (526 bytes) to UDP:123.253.126.205:44354 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.69.139:44354;rport=44354;received=123.253.126.205;branch=z9hG4bK-524287-1---e25ffcb5ce799478
Call-ID: vVWiZBwt0yMGTPU0nbFKWw..
From: <sip:7001@13.200.0.116>;tag=7b820379
To: <sip:8144098500@13.200.0.116>;tag=z9hG4bK-524287-1---e25ffcb5ce799478
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1694430264/63f32e1b1319c0ea75e3f4a4d979aa1b",opaque="718acf2051a31708",algorithm=md5,qop="auth"
Server: Asterisk PBX certified-18.9-cert5
Content-Length: 0

<--- Received SIP request (373 bytes) from UDP:123.253.126.205:44354 --->
ACK sip:8144098500@13.200.0.116:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.69.139:44354;branch=z9hG4bK-524287-1---e25ffcb5ce799478;rport
Max-Forwards: 70
To: <sip:8144098500@13.200.0.116>;tag=z9hG4bK-524287-1---e25ffcb5ce799478
From: <sip:7001@13.200.0.116:5060;transport=UDP>;tag=7b820379
Call-ID: vVWiZBwt0yMGTPU0nbFKWw..
CSeq: 1 ACK
Content-Length: 0

<--- Received SIP request (1360 bytes) from UDP:123.253.126.205:44354 --->
INVITE sip:8144098500@13.200.0.116:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.69.139:44354;branch=z9hG4bK-524287-1---377dbbfb69a75817;rport
Max-Forwards: 70
Contact: <sip:7001@123.253.126.205:44354;transport=UDP>
To: <sip:8144098500@13.200.0.116:5060>
From: <sip:7001@13.200.0.116:5060;transport=UDP>;tag=7b820379
Call-ID: vVWiZBwt0yMGTPU0nbFKWw..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Authorization: Digest username="7001",realm="asterisk",nonce="1694430264/63f32e1b1319c0ea75e3f4a4d979aa1b",uri="sip:8144098500@13.200.0.116:5060;transport=UDP",response="0571dba7850107aef08c66354f0d9590",cnonce="efb965dfac43486314d2de466ce71a34",nc=00000001,qop=auth,algorithm=md5,opaque="718acf2051a31708"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 346

v=0
o=Z 0 21006532 IN IP4 123.253.126.205
s=Z
c=IN IP4 123.253.126.205
t=0 0
m=audio 32816 RTP/AVP 106 9 98 101 0 8 3


a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP response (334 bytes) to UDP:123.253.126.205:44354 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.69.139:44354;rport=44354;received=123.253.126.205;branch=z9hG4bK-524287-1---377dbbfb69a75817
Call-ID: vVWiZBwt0yMGTPU0nbFKWw..
From: <sip:7001@13.200.0.116>;tag=7b820379
To: <sip:8144098500@13.200.0.116>
CSeq: 2 INVITE
Server: Asterisk PBX certified-18.9-cert5
Content-Length: 0

-- Executing [8144098500@from-internal:1] Set("PJSIP/7001-00000014", "CALLERID(all)="Bob" <+15551224722>") in new stack
-- Executing [8144098500@from-internal:2] Dial("PJSIP/7001-00000014", "PJSIP/+18144098500@VoIPVoIP-endpoint") in new stack
-- Called PJSIP/+18144098500@VoIPVoIP-endpoint

<--- Transmitting SIP request (1074 bytes) to UDP:66.220.10.66:5060 --->
INVITE sip:sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 13.200.0.116:5060;rport;branch=z9hG4bKPjac15feb8-ed72-4dfe-8139-d05c21c0a739
From: <sip:5551231234@sip3.voipvoip.com>;tag=64729f4c-a5e4-42ca-b73a-c217606d02f1
To: <sip:+18144098500@sip3.voipvoip.com>
Contact: <sip:5551231234@13.200.0.116:5060>
Call-ID: 6c6fb020-086e-4396-8598-60dbad3ac5a3
CSeq: 15165 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Route: <sip:+18144098500@sip3.voipvoip.com>
Max-Forwards: 70
User-Agent: Asterisk PBX certified-18.9-cert5
Content-Type: application/sdp
Content-Length: 334

v=0
o=- 1569496270 1569496270 IN IP4 113.200.0.116
s=Asterisk
c=IN IP4 113.200.0.116
t=0 0
m=audio 15434 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (527 bytes) from UDP:66.220.10.66:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 13.200.0.116:5060;received=13.200.0.116;rport=5060;branch=z9hG4bKPjac15feb8-ed72-4dfe-8139-d05c21c0a739
From: <sip:5551231234@sip3.voipvoip.com>;tag=64729f4c-a5e4-42ca-b73a-c217606d02f1
To: <sip:+18144098500@sip3.voipvoip.com>;tag=2db53d739291a55e2d6095d250d30e81.5f18
Call-ID: 6c6fb020-086e-4396-8598-60dbad3ac5a3
CSeq: 15165 INVITE
Proxy-Authenticate: Digest realm="voipvoip.com", nonce="64fef567000137b244636745c608241d65999cd36eee0722"
Content-Length: 0

<--- Transmitting SIP request (493 bytes) to UDP:66.220.10.66:5060 --->
ACK sip:sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 13.200.0.116:5060;rport;branch=z9hG4bKPjac15feb8-ed72-4dfe-8139-d05c21c0a739
From: <sip:5551231234@sip3.voipvoip.com>;tag=64729f4c-a5e4-42ca-b73a-c217606d02f1
To: <sip:+18144098500@sip3.voipvoip.com>;tag=2db53d739291a55e2d6095d250d30e81.5f18
Call-ID: 6c6fb020-086e-4396-8598-60dbad3ac5a3
CSeq: 15165 ACK
Route: <sip:+18144098500@sip3.voipvoip.com>
Max-Forwards: 70
User-Agent: Asterisk PBX certified-18.9-cert5
Content-Length: 0

<--- Transmitting SIP request (1279 bytes) to UDP:66.220.10.66:5060 --->
INVITE sip:sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 13.200.0.116:5060;rport;branch=z9hG4bKPjf7d7e89b-9628-4752-b706-9ffde7b5400a
From

: <sip:5551231234@sip3.voipvoip.com>;tag=64729f4c-a5e4-42ca-b73a-c217606d02f1
To: <sip:+18144098500@sip3.voipvoip.com>
Contact: <sip:5551231234@13.200.0.116:5060>
Call-ID: 6c6fb020-086e-4396-8598-60dbad3ac5a3
CSeq: 15166 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX certified-18.9-cert5
Proxy-Authorization: Digest username="5551231234", realm="voipvoip.com", nonce="64fef567000137b244636745c608241d65999cd36eee0722", uri="sip:sip3.voipvoip.com", response="96075069a3a2ec9509f4c4fe7dea8088"
Route: <sip:+18144098500@sip3.voipvoip.com>
Content-Type: application/sdp
Content-Length: 334

v=0
o=- 1569496270 1569496270 IN IP4 113.200.0.116
s=Asterisk
c=IN IP4 113.200.0.116
t=0 0
m=audio 15434 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (413 bytes) from UDP:66.220.10.66:5060 --->
SIP/2.0 484 Address not acceptable
Via: SIP/2.0/UDP 13.200.0.116;received=13.200.0.116;rport=5060;branch=z9hG4bKPjf7d7e89b-9628-4752-b706-9ffde7b5400a
From: <sip:5551231234@sip3.voipvoip.com>;tag=64729f4c-a5e4-42ca-b73a-c217606d02f1
To: <sip:+18144098500@sip3.voipvoip.com>;tag=c1de1eb30f0a9bcda464e4f5339306cc-45d8
Call-ID: 6c6fb020-086e-4396-8598-60dbad3ac5a3
CSeq: 15166 INVITE
Content-Length: 0

<--- Transmitting SIP request (493 bytes) to UDP:66.220.10.66:5060 --->
ACK sip:sip3.voipvoip.com SIP/2.0
Via: SIP/2.0/UDP 13.200.0.116:5060;rport;branch=z9hG4bKPjf7d7e89b-9628-4752-b706-9ffde7b5400a
From: <sip:5551231234@sip3.voipvoip.com>;tag=64729f4c-a5e4-42ca-b73a-c217606d02f1
To: <sip:+18144098500@sip3.voipvoip.com>;tag=c1de1eb30f0a9bcda464e4f5339306cc-45d8
Call-ID: 6c6fb020-086e-4396-8598-60dbad3ac5a3
CSeq: 15166 ACK
Route: <sip:+18144098500@sip3.voipvoip.com>
Max-Forwards: 70
User-Agent: Asterisk PBX certified-18.9-cert5
Content-Length: 0

== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [8144098500@from-internal:3] Hangup("PJSIP/7001-00000014", "") in new stack
== Spawn extension (from-internal, 8144098500, 3) exited non-zero on 'PJSIP/7001-00000014'

<--- Transmitting SIP response (411 bytes) to UDP:123.253.126.205:44354 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.69.139:44354;rport=44354;received=123.253.126.205;branch=z9hG4bK-524287-1---377dbbfb69a75817
Call-ID: vVWiZBwt0yMGTPU0nbFKWw..
From: <sip:7001@13.200.0.116>;tag=7b820379
To: <sip:8144098500@13.200.0.116>;tag=fc3b4951-bf12-4c2b-8535-72b4dbe634e4
CSeq: 2 INVITE
Server: Asterisk PBX certified-18.9-cert5
Reason: Q.850;cause=28
Content-Length: 0

<--- Received SIP request (374 bytes) from UDP:123.253.126.205:44354 --->
ACK sip:8144098500@13.200.0.116:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.69.139:44354;branch=z9hG4bK-524287-1---377dbbfb69a75817;rport
Max-Forwards: 70
To: <sip:8144098500@13.200.0.116>;tag=fc3b4951-bf12-4c2b-8535-72b4dbe634e4
From: <sip:7001@13.200.0.116:5060;transport=UDP>;tag=7b820379
Call-ID: vVWiZBwt0yMGTPU0nbFKWw..
CSeq: 2 ACK
Content-Length: 0

And the registration status output from asterisk console:

ip-172-31-41-232*CLI> pjsip show endpoints

 Endpoint:  <Endpoint/CID.....................................>  <State.....>  <Channels.>
    I/OAuth:  <AuthId/UserName...........................................................>
        Aor:  <Aor............................................>  <MaxContact>
      Contact:  <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
  Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress..................>
   Identify:  <Identify/Endpoint.........................................................>
        Match:  <criteria.........................>
    Channel:  <ChannelId......................................>  <State.....>  <Time.....>
        Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
==========================================================================================

 Endpoint:  7001/7001                                            Not in use    0 of inf
     InAuth:  7001/7001
        Aor:  7001                                               2
      Contact:  7001/sip:7001@123.253.126.205:44354;transp bd960e73bc NonQual         nan
  Transport:  0.0.0.0-udp               udp      3     96  0.0.0.0:5060



 Endpoint:  VoIPVoIP-endpoint                                    Not in use    0 of inf
    OutAuth:  VoIPVoIP-auth/5551231234
     InAuth:  VoIPVoIP-auth/5551231234
        Aor:  VoIPVoIP-aor                                       0
      Contact:  VoIPVoIP-aor/sip:sip3.voipvoip.com         e65a5f666e NonQual         nan
  Transport:  0.0.0.0-udp               udp      3     96  0.0.0.0:5060
   Identify:  VoIPVoIP-identify/VoIPVoIP-endpoint
        Match: 66.220.10.66/32


Objects found: 3


==========================================================================================


ip-172-31-41-232*CLI> pjsip show registrations

 <Registration/ServerURI..............................>  <Auth..........>  <Status.......>
==========================================================================================

 VoIPVoIP-registration/sip:sip3.voipvoip.com             VoIPVoIP-auth     Registered      

Objects found: 1
7

You’ll want to remove the outbound_proxy setting from the endpoint. It serves no purpose and is your issue.

8

@jcolp

Thank you so much ! that was the issue , but when i call other numbers to test , a few go through successfully and other throw the same error and the successful ones when answer the call there’s no audio whatsoever. could you please help with that what could be the problem?

I tried changing codecs (allowing g729,711 etc. ) tried different softphones.

it’d be of great help if you could point me in a direction , where to look.

9

The provider claims not to like the number you are calling, although it is possible they are re-purposing the message for something else.

If the number is intended to be an obfuscated NANP one, I’d suggest that they were expecting 1555… or +1555…

10

Hey @david551 ,

I tried what @jcolp suggested , removing outbound_proxy solved the issue with “busy/congested” error but there’s no audio whatsoever

11

“No audio” is a separate issue than your OP of “busy/congested” and should probably be in a separate thread because now that you marked this post as “Solved”, it won’t get the same level of attention.

1 Like
12

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