Problem with outbound PJSIP - Everyone is busy/congested at this time

Vince_be · October 11, 2023, 8:22pm

Hi all,

Progressing slowly in migration from Asterisk 16 SIP to Asterisk 20 PJSIP.

Inbound calls are ok now (2 numbers, 1 registration).

But when trying outbound calls, I get the message : “Everyone is busy/congested at this time”.

Any idea ?

Thanks in advance !

pjsip.conf

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:56278
local_net=172.XX.XX.XX/255.255.240.0
external_signaling_address=13.XX.XX.XX
external_media_address=13.XX.XX.XX

[mytrunk_reg]
type=registration
transport=transport-udp
outbound_auth=mytrunk_auth
server_uri=sip:voip.mytrunk.com
client_uri=sip:user@voip.mytrunk.com
contact_user=user
retry_interval=60
forbidden_retry_interval=600
expiration=3600
line=yes
endpoint=mytrunk_edp

[mytrunk_auth]
type=auth
auth_type=userpass
password=password
username=user
realm=voip.mytrunk.com

[mytrunk_edp]
type=endpoint
transport=transport-udp
context=from-external
disallow=all
allow=ulaw
outbound_auth=mytrunk_auth
aors=mytrunk_aor
force_rport=yes
direct_media=no

[mytrunk_aor]
type=aor
contact=sip:188.XX.XX.XX:5060
contact=sip:188.XX.XX.XX:6060

[mytrunk_ide]
type=identify
endpoint=mytrunk_edp
match=188.XX.XX.XX:5060
match=188.XX.XX.XX:6060

extensions.conf :

exten => _0.,1,NoOp([phone to mytrunk])
exten => _0.,n,Set(CALLERID(all)="Test <3280XXXXXX>"}) 
exten => _0.,n,Dial(PJSIP/${EXTEN}@mytrunk_edp)
exten => _0.,n,NoOp(Dial status : ${DIALSTATUS}) 
exten => _0.,n,Hangup()

CLI output :

    -- Executing [04XXXXXXXX@from-internal:1] NoOp("PJSIP/phone2-00000000", "[phone to mytrunk]") in new stack
    -- Executing [04XXXXXXXX@from-internal:2] Set("PJSIP/phone2-00000000", "CALLERID(all)="Test <3280XXXXXX>"}") in new stack
    -- Executing [04XXXXXXXX@from-internal:3] Dial("PJSIP/phone2-00000000", "PJSIP/04XXXXXXXX@mytrunk_edp") in new stack
    -- Called PJSIP/04XXXXXXXX@mytrunk_edp
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [04XXXXXXXX@from-internal:4] NoOp("PJSIP/phone2-00000000", "Dial status : CHANUNAVAIL") in new stack
    -- Executing [04XXXXXXXX@from-internal:5] Hangup("PJSIP/phone2-00000000", "") in new stack
  == Spawn extension (from-internal, 04XXXXXXXX, 5) exited non-zero on 'PJSIP/phone2-00000000'
asterisk*CLI>

PJSIP log :

<--- Received SIP request (915 bytes) from UDP:91.XX.XX.XX:57809 --->
INVITE sip:0470XXXXXX@13.XX.XX.XX:56278 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.110:57809;branch=z9hG4bK-524287-1---2c70c67ef9bf0d34;rport
Max-Forwards: 70
Contact: <sip:phone2@192.168.2.110:57809>
To: <sip:0470XXXXXX@13.XX.XX.XX:56278>
From: "phone2"<sip:phone2@13.XX.XX.XX:56278>;tag=84c4d50c
Call-ID: 5SofgU6J8cprYkxDsSnq7g..
CSeq: 1 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Content-Type: application/sdp
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 6.1
Content-Length: 304

v=0
o=- 0 1 IN IP4 192.168.0.250
s=-
c=IN IP4 192.168.2.110
t=0 0
m=audio 4016 RTP/AVP 3 102 0 8 9 101
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP response (525 bytes) to UDP:91.XX.XX.XX:57809 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.110:57809;rport=57809;received=91.XX.XX.XX;branch=z9hG4bK-524287-1---2c70c67ef9bf0d34
Call-ID: 5SofgU6J8cprYkxDsSnq7g..
From: "phone2" <sip:phone2@13.XX.XX.XX>;tag=84c4d50c
To: <sip:0470XXXXXX@13.XX.XX.XX>;tag=z9hG4bK-524287-1---2c70c67ef9bf0d34
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1697054907/bd7c98ce3a9b544c14088a52ea1f96a4",opaque="1d28421b6697364f",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.4.0
Content-Length:  0


<--- Received SIP request (360 bytes) from UDP:91.XX.XX.XX:57809 --->
ACK sip:0470XXXXXX@13.XX.XX.XX:56278 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.110:57809;branch=z9hG4bK-524287-1---2c70c67ef9bf0d34;rport
Max-Forwards: 70
To: <sip:0470XXXXXX@13.XX.XX.XX>;tag=z9hG4bK-524287-1---2c70c67ef9bf0d34
From: "phone2"<sip:phone2@13.XX.XX.XX:56278>;tag=84c4d50c
Call-ID: 5SofgU6J8cprYkxDsSnq7g..
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1214 bytes) from UDP:91.XX.XX.XX:57809 --->
INVITE sip:0470XXXXXX@13.XX.XX.XX:56278 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.110:57809;branch=z9hG4bK-524287-1---1ec353107fb2b22b;rport
Max-Forwards: 70
Contact: <sip:phone2@192.168.2.110:57809>
To: <sip:0470XXXXXX@13.XX.XX.XX:56278>
From: "phone2"<sip:phone2@13.XX.XX.XX:56278>;tag=84c4d50c
Call-ID: 5SofgU6J8cprYkxDsSnq7g..
CSeq: 2 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Content-Type: application/sdp
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 6.1
Authorization: Digest username="phone2",realm="asterisk",nonce="1697054907/bd7c98ce3a9b544c14088a52ea1f96a4",uri="sip:0470XXXXXX@13.XX.XX.XX:56278",response="0033526f1324ac3c2b5a97d076bc7385",cnonce="61eee907db278f4a608d704951baba42",nc=00000001,qop=auth,algorithm=MD5,opaque="1d28421b6697364f"
Content-Length: 304

v=0
o=- 0 1 IN IP4 192.168.0.250
s=-
c=IN IP4 192.168.2.110
t=0 0
m=audio 4016 RTP/AVP 3 102 0 8 9 101
a=rtpmap:3 GSM/8000
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP response (333 bytes) to UDP:91.XX.XX.XX:57809 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.110:57809;rport=57809;received=91.XX.XX.XX;branch=z9hG4bK-524287-1---1ec353107fb2b22b
Call-ID: 5SofgU6J8cprYkxDsSnq7g..
From: "phone2" <sip:phone2@13.XX.XX.XX>;tag=84c4d50c
To: <sip:0470XXXXXX@13.XX.XX.XX>
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Content-Length:  0


    -- Executing [0470XXXXXX@from-internal:1] NoOp("PJSIP/phone2-00000002", "[phone to mytrunk]") in new stack
    -- Executing [0470XXXXXX@from-internal:2] Set("PJSIP/phone2-00000002", "CALLERID(all)="Test <3280XXXXXX>"}") in new stack
    -- Executing [0470XXXXXX@from-internal:3] Dial("PJSIP/phone2-00000002", "PJSIP/0470XXXXXX@3starsnet_edp") in new stack
    -- Called PJSIP/0470XXXXXX@3starsnet_edp
<--- Transmitting SIP request (930 bytes) to UDP:188.XX.XX.XX:6060 --->
INVITE sip:0470XXXXXX@188.XX.XX.XX:6060 SIP/2.0
Via: SIP/2.0/UDP 13.XX.XX.XX:56278;rport;branch=z9hG4bKPjc0084d6a-5f86-4e3f-b34f-7ee8d67454d9
From: "\"Test" <sip:3280XXXXXX@172.31.47.240>;tag=12a43866-1164-46f4-8935-2ce6e5054473
To: <sip:0470XXXXXX@188.XX.XX.XX>
Contact: <sip:asterisk@13.XX.XX.XX:56278>
Call-ID: 77886ffd-3163-43f3-88f9-f6aa65015b82
CSeq: 11338 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 20.4.0
Content-Type: application/sdp
Content-Length:   235

v=0
o=- 809177479 809177479 IN IP4 13.XX.XX.XX
s=Asterisk
c=IN IP4 13.XX.XX.XX
t=0 0
m=audio 15516 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (421 bytes) from UDP:188.XX.XX.XX:6060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 13.XX.XX.XX:56278;rport=56278;branch=z9hG4bKPjc0084d6a-5f86-4e3f-b34f-7ee8d67454d9;received=13.XX.XX.XX
From: "\"Test" <sip:3280XXXXXX@172.31.47.240>;tag=12a43866-1164-46f4-8935-2ce6e5054473
To: <sip:0470XXXXXX@188.XX.XX.XX>
Call-ID: 77886ffd-3163-43f3-88f9-f6aa65015b82
CSeq: 11338 INVITE
Server: Enswitch SIP proxy
Content-Length: 0


<--- Received SIP response (565 bytes) from UDP:188.XX.XX.XX:6060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 13.XX.XX.XX:56278;received=13.XX.XX.XX;rport=56278;branch=z9hG4bKPjc0084d6a-5f86-4e3f-b34f-7ee8d67454d9
From: "\"Test" <sip:3280XXXXXX@172.31.47.240>;tag=12a43866-1164-46f4-8935-2ce6e5054473
To: <sip:0470XXXXXX@188.XX.XX.XX>;tag=as466ea858
Call-ID: 77886ffd-3163-43f3-88f9-f6aa65015b82
CSeq: 11338 INVITE
Server: mytrunk VoipSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0


<--- Transmitting SIP request (422 bytes) to UDP:188.XX.XX.XX:6060 --->
ACK sip:0470XXXXXX@188.XX.XX.XX:6060 SIP/2.0
Via: SIP/2.0/UDP 13.XX.XX.XX:56278;rport;branch=z9hG4bKPjc0084d6a-5f86-4e3f-b34f-7ee8d67454d9
From: "\"Test" <sip:3280XXXXXX@172.31.47.240>;tag=12a43866-1164-46f4-8935-2ce6e5054473
To: <sip:0470XXXXXX@188.XX.XX.XX>;tag=as466ea858
Call-ID: 77886ffd-3163-43f3-88f9-f6aa65015b82
CSeq: 11338 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.4.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [0470XXXXXX@from-internal:4] NoOp("PJSIP/phone2-00000002", "Dial status : CHANUNAVAIL") in new stack
    -- Executing [0470XXXXXX@from-internal:5] Hangup("PJSIP/phone2-00000002", "") in new stack
  == Spawn extension (from-internal, 0470XXXXXX, 5) exited non-zero on 'PJSIP/phone2-00000002'
<--- Transmitting SIP response (401 bytes) to UDP:91.XX.XX.XX:57809 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.2.110:57809;rport=57809;received=91.XX.XX.XX;branch=z9hG4bK-524287-1---1ec353107fb2b22b
Call-ID: 5SofgU6J8cprYkxDsSnq7g..
From: "phone2" <sip:phone2@13.XX.XX.XX>;tag=84c4d50c
To: <sip:0470XXXXXX@13.XX.XX.XX>;tag=eb887d85-c070-41a1-807f-57124d890d45
CSeq: 2 INVITE
Server: Asterisk PBX 20.4.0
Reason: Q.850;cause=21
Content-Length:  0


<--- Received SIP request (361 bytes) from UDP:91.XX.XX.XX:57809 --->
ACK sip:0470XXXXXX@13.XX.XX.XX:56278 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.110:57809;branch=z9hG4bK-524287-1---1ec353107fb2b22b;rport
Max-Forwards: 70
To: <sip:0470XXXXXX@13.XX.XX.XX>;tag=eb887d85-c070-41a1-807f-57124d890d45
From: "phone2"<sip:phone2@13.XX.XX.XX:56278>;tag=84c4d50c
Call-ID: 5SofgU6J8cprYkxDsSnq7g..
CSeq: 2 ACK
Content-Length: 0


asterisk*CLI>

david551 · October 11, 2023, 8:44pm

The destination has refused your call, typically because it violates some terms of use. You will need to ask whoever maintains it.

Note the headline message is a secondary one, and the important part is (0,0,1) which tells you the reason is unavailable, rather than busy or congested.

tootai · October 11, 2023, 8:56pm

Wrong exten => _0.,n,Set(CALLERID(all)=“Test <3280XXXXXX>”})

should be

exten => _0.,n,Set(CALLERID(all)="Test <3280XXXXXX>")

Vince_be · October 11, 2023, 9:01pm

Thanks David. I will contact my provider.

Vince_be · October 11, 2023, 9:12pm

Corrected. Thanks!

Vince_be · October 12, 2023, 12:44pm

I contacted the provider and he mentioned that I would miss a identify_by=username in the definition of my endpoint for outbound calls?

Does that ring a bell to you?

Thanks in advance

david551 · October 12, 2023, 1:40pm

That affects inbound, not outbound. and would be unusual on the provider side.

Vince_be · October 13, 2023, 4:48pm

Hi all,

I found the issue. The statement : “from_domain=” was missing in the definition of the endpoint.

I hope it helps others

Regards

system · November 12, 2023, 4:49pm

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

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