Outbound getting error this number is not yet assigned

here’s a new update

[2023-02-25 12:37:22] VERBOSE[794140] res_pjsip_logger.c: <— Received SIP request (895 bytes) from UDP:192.168.0.14:37553 —>
INVITE sip:4384082316@192.168.0.14;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:37553;branch=z9hG4bK-524287-1—d676ebda8fc2d34c;rport
Max-Forwards: 70
Contact: sip:1001@192.168.0.14:37553;transport=UDP
To: sip:4384082316@192.168.0.14
From: sip:1001@192.168.0.14;transport=UDP;tag=f5bf0e2a
Call-ID: WcWyJ2BblsrFTAadMq5esg…
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.10 v2.10.17.3
Allow-Events: presence, kpml, talk
Content-Length: 329

v=0
o=Z 589229602 1 IN IP4 192.168.0.14
s=Z
c=IN IP4 192.168.0.14
t=0 0
m=audio 45312 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

[2023-02-25 12:37:22] VERBOSE[794141] res_pjsip_logger.c: <— Transmitting SIP response (518 bytes) to UDP:192.168.0.14:37553 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.14:37553;rport=37553;received=192.168.0.14;branch=z9hG4bK-524287-1—d676ebda8fc2d34c
Call-ID: WcWyJ2BblsrFTAadMq5esg…
From: sip:1001@192.168.0.14;tag=f5bf0e2a
To: sip:4384082316@192.168.0.14;tag=z9hG4bK-524287-1—d676ebda8fc2d34c
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1677346642/0744f0ae201c4e494c19adeb787f8d02”,opaque=“7c75dc664b3ac21d”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX GIT-18-c5c858287a
Content-Length: 0

[2023-02-25 12:37:22] VERBOSE[794140] res_pjsip_logger.c: <— Received SIP request (361 bytes) from UDP:192.168.0.14:37553 —>
ACK sip:4384082316@192.168.0.14;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:37553;branch=z9hG4bK-524287-1—d676ebda8fc2d34c;rport
Max-Forwards: 70
To: sip:4384082316@192.168.0.14;tag=z9hG4bK-524287-1—d676ebda8fc2d34c
From: sip:1001@192.168.0.14;transport=UDP;tag=f5bf0e2a
Call-ID: WcWyJ2BblsrFTAadMq5esg…
CSeq: 1 ACK
Content-Length: 0

[2023-02-25 12:37:22] VERBOSE[794140] res_pjsip_logger.c: <— Received SIP request (1198 bytes) from UDP:192.168.0.14:37553 —>
INVITE sip:4384082316@192.168.0.14;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:37553;branch=z9hG4bK-524287-1—07cbb56a1dabab89;rport
Max-Forwards: 70
Contact: sip:1001@192.168.0.14:37553;transport=UDP
To: sip:4384082316@192.168.0.14
From: sip:1001@192.168.0.14;transport=UDP;tag=f5bf0e2a
Call-ID: WcWyJ2BblsrFTAadMq5esg…
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.10 v2.10.17.3
Authorization: Digest username=“1001”,realm=“asterisk”,nonce=“1677346642/0744f0ae201c4e494c19adeb787f8d02”,uri="sip:4384082316@192.168.0.14;transport=UDP",response=“f9deedd4175801aeb206541371d5e593”,cnonce=“4f7ca966e261b2bd69fac5e10bf3f1ce”,nc=00000001,qop=auth,algorithm=MD5,opaque=“7c75dc664b3ac21d”
Allow-Events: presence, kpml, talk
Content-Length: 329

v=0
o=Z 589229602 1 IN IP4 192.168.0.14
s=Z
c=IN IP4 192.168.0.14
t=0 0
m=audio 45312 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

[2023-02-25 12:37:22] VERBOSE[794141] res_pjsip_logger.c: <— Transmitting SIP response (326 bytes) to UDP:192.168.0.14:37553 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.14:37553;rport=37553;received=192.168.0.14;branch=z9hG4bK-524287-1—07cbb56a1dabab89
Call-ID: WcWyJ2BblsrFTAadMq5esg…
From: sip:1001@192.168.0.14;tag=f5bf0e2a
To: sip:4384082316@192.168.0.14
CSeq: 2 INVITE
Server: Asterisk PBX GIT-18-c5c858287a
Content-Length: 0

[2023-02-25 12:37:22] VERBOSE[795068][C-00000008] pbx_realtime.c: Executing [4384082316@internal:1] Dial(“PJSIP/1001-0000000e”, “PJSIP/4384082316@voipms”)
[2023-02-25 12:37:22] VERBOSE[795068][C-00000008] app_dial.c: Called PJSIP/4384082316@voipms
[2023-02-25 12:37:22] VERBOSE[794141] res_pjsip_logger.c: <— Transmitting SIP request (1049 bytes) to UDP:208.100.60.17:5060 —>
INVITE sip:4384082316@atlanta.voip.ms SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;rport;branch=z9hG4bKPj7b6e98f3-4caa-41ac-bad4-328a16c7e9aa
From: sip:hiding_username@192.168.0.14;tag=f623ab8e-c31c-4469-ba12-92a6445e2a2b
To: sip:4384082316@atlanta.voip.ms
Contact: sip:hiding_username@192.168.0.14:5060
Call-ID: 2e713889-e7b9-49d9-9b35-c1cc90906bf1
CSeq: 9995 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
Remote-Party-ID: sip:1001@192.168.0.14;party=calling;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-c5c858287a
Content-Type: application/sdp
Content-Length: 282

v=0
o=- 820355014 820355014 IN IP4 192.168.0.14
s=Asterisk
c=IN IP4 192.168.0.14
t=0 0
m=audio 11652 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 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2023-02-25 12:37:22] VERBOSE[794140] res_pjsip_logger.c: <— Received SIP response (590 bytes) from UDP:208.100.60.17:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bKPj7b6e98f3-4caa-41ac-bad4-328a16c7e9aa;received=192.168.0.14;rport=5060
From: sip:hiding_username@192.168.0.14:5060;tag=f623ab8e-c31c-4469-ba12-92a6445e2a2b
To: sip:4384082316@atlanta.voip.ms;tag=as6935e319
Call-ID: 2e713889-e7b9-49d9-9b35-c1cc90906bf1
CSeq: 9995 INVITE
Server: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“atlanta.voip.ms”, nonce=“756d25b5”
Content-Length: 0

[2023-02-25 12:37:22] VERBOSE[794141] res_pjsip_logger.c: <— Transmitting SIP request (416 bytes) to UDP:208.100.60.17:5060 —>
ACK sip:4384082316@atlanta.voip.ms SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;rport;branch=z9hG4bKPj7b6e98f3-4caa-41ac-bad4-328a16c7e9aa
From: sip:hiding_username@192.168.0.14;tag=f623ab8e-c31c-4469-ba12-92a6445e2a2b
To: sip:4384082316@atlanta.voip.ms;tag=as6935e319
Call-ID: 2e713889-e7b9-49d9-9b35-c1cc90906bf1
CSeq: 9995 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-c5c858287a
Content-Length: 0

[2023-02-25 12:37:22] VERBOSE[794141] res_pjsip_logger.c: <— Transmitting SIP request (1234 bytes) to UDP:208.100.60.17:5060 —>
INVITE sip:4384082316@atlanta.voip.ms SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;rport;branch=z9hG4bKPj13b1e513-70e2-4e8f-a243-e06f5efeb8e5
From: sip:hiding_username@192.168.0.14;tag=f623ab8e-c31c-4469-ba12-92a6445e2a2b
To: sip:4384082316@atlanta.voip.ms
Contact: sip:hiding_username@192.168.0.14:5060
Call-ID: 2e713889-e7b9-49d9-9b35-c1cc90906bf1
CSeq: 9996 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 GIT-18-c5c858287a
Authorization: Digest username=“hiding_username”, realm=“atlanta.voip.ms”, nonce=“756d25b5”, uri=“sip:4384082316@atlanta.voip.ms”, response=“3ffa4cb8593851f5fa481f3dcf809353”, algorithm=MD5
Remote-Party-ID: sip:1001@192.168.0.14;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 282

v=0
o=- 820355014 820355014 IN IP4 192.168.0.14
s=Asterisk
c=IN IP4 192.168.0.14
t=0 0
m=audio 11652 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 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2023-02-25 12:37:23] VERBOSE[794140] res_pjsip_logger.c: <— Received SIP response (569 bytes) from UDP:208.100.60.17:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bKPj13b1e513-70e2-4e8f-a243-e06f5efeb8e5;received=192.168.0.14;rport=5060
From: sip:hiding_username@192.168.0.14:5060;tag=f623ab8e-c31c-4469-ba12-92a6445e2a2b
To: sip:4384082316@atlanta.voip.ms
Call-ID: 2e713889-e7b9-49d9-9b35-c1cc90906bf1
CSeq: 9996 INVITE
Server: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:4384082316@208.100.60.17:5060
Content-Length: 0

[2023-02-25 12:37:23] VERBOSE[794140] res_pjsip_logger.c: <— Received SIP response (551 bytes) from UDP:208.100.60.17:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bKPj13b1e513-70e2-4e8f-a243-e06f5efeb8e5;received=192.168.0.14;rport=5060
From: sip:hiding_username@192.168.0.14:5060;tag=f623ab8e-c31c-4469-ba12-92a6445e2a2b
To: sip:4384082316@atlanta.voip.ms;tag=as32365845
Call-ID: 2e713889-e7b9-49d9-9b35-c1cc90906bf1
CSeq: 9996 INVITE
Server: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0

[2023-02-25 12:37:23] VERBOSE[794141] res_pjsip_logger.c: <— Transmitting SIP request (416 bytes) to UDP:208.100.60.17:5060 —>
ACK sip:4384082316@atlanta.voip.ms SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;rport;branch=z9hG4bKPj13b1e513-70e2-4e8f-a243-e06f5efeb8e5
From: sip:hiding_username@192.168.0.14;tag=f623ab8e-c31c-4469-ba12-92a6445e2a2b
To: sip:4384082316@atlanta.voip.ms;tag=as32365845
Call-ID: 2e713889-e7b9-49d9-9b35-c1cc90906bf1
CSeq: 9996 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-c5c858287a
Content-Length: 0

[2023-02-25 12:37:23] VERBOSE[795068][C-00000008] app_dial.c: Everyone is busy/congested at this time (1:0/1/0)
[2023-02-25 12:37:23] VERBOSE[795068][C-00000008] pbx.c: Auto fallthrough, channel ‘PJSIP/1001-0000000e’ status is ‘CONGESTION’
[2023-02-25 12:37:23] VERBOSE[794141] res_pjsip_logger.c: <— Transmitting SIP response (404 bytes) to UDP:192.168.0.14:37553 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.14:37553;rport=37553;received=192.168.0.14;branch=z9hG4bK-524287-1—07cbb56a1dabab89
Call-ID: WcWyJ2BblsrFTAadMq5esg…
From: sip:1001@192.168.0.14;tag=f5bf0e2a
To: sip:4384082316@192.168.0.14;tag=d8f43656-93f3-4dcd-8026-4da4b6f09c52
CSeq: 2 INVITE
Server: Asterisk PBX GIT-18-c5c858287a
Reason: Q.850;cause=34
Content-Length: 0

[2023-02-25 12:37:23] VERBOSE[794140] res_pjsip_logger.c: <— Received SIP request (362 bytes) from UDP:192.168.0.14:37553 —>
ACK sip:4384082316@192.168.0.14;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:37553;branch=z9hG4bK-524287-1—07cbb56a1dabab89;rport
Max-Forwards: 70
To: sip:4384082316@192.168.0.14;tag=d8f43656-93f3-4dcd-8026-4da4b6f09c52
From: sip:1001@192.168.0.14;transport=UDP;tag=f5bf0e2a
Call-ID: WcWyJ2BblsrFTAadMq5esg…
CSeq: 2 ACK
Content-Length: 0

[2023-02-25 12:37:28] VERBOSE[794141] res_pjsip_logger.c: <— Transmitting SIP request (500 bytes) to UDP:192.168.0.14:37553 —>
OPTIONS sip:1001@192.168.0.14:37553;transport=UDP;rinstance=ce61bbdeb1fbec2e SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;rport;branch=z9hG4bKPj302a314f-cc93-4c14-a388-2db28f0a67b1
From: sip:1001@192.168.0.14;tag=5b727b27-6203-4ffb-bff0-989216f4522d
To: sip:1001@192.168.0.14;rinstance=ce61bbdeb1fbec2e
Contact: sip:1001@192.168.0.14:5060
Call-ID: db3cb565-7ca4-4206-8535-9a1d362f184c
CSeq: 34684 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-c5c858287a
Content-Length: 0

[2023-02-25 12:37:28] VERBOSE[794140] res_pjsip_logger.c: <— Received SIP response (688 bytes) from UDP:192.168.0.14:37553 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5060;rport=5060;branch=z9hG4bKPj302a314f-cc93-4c14-a388-2db28f0a67b1
Contact: sip:192.168.0.14:37553
To: sip:1001@192.168.0.14;rinstance=ce61bbdeb1fbec2e;tag=06685f5f
From: sip:1001@192.168.0.14;tag=5b727b27-6203-4ffb-bff0-989216f4522d
Call-ID: db3cb565-7ca4-4206-8535-9a1d362f184c
CSeq: 34684 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.5.10 v2.10.17.3
Allow-Events: presence, kpml, talk
Content-Length: 0

[2023-02-25 12:37:34] VERBOSE[794141] res_pjsip_logger.c: <— Transmitting SIP request (451 bytes) to UDP:208.100.60.17:5060 —>
OPTIONS sip:hiding_username@atlanta.voip.ms SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5060;rport;branch=z9hG4bKPj1b6b58fd-28f1-4fed-9c6f-be332e97572b
From: sip:hiding_username@192.168.0.14;tag=9318364c-ec29-4302-ab4d-f2266ef948be
To: sip:hiding_username@atlanta.voip.ms
Contact: sip:hiding_username@192.168.0.14:5060
Call-ID: 8951857f-8080-47fe-a9e6-636c0cc4abbf
CSeq: 1646 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-18-c5c858287a
Content-Length: 0

[2023-02-25 12:37:34] VERBOSE[794140] res_pjsip_logger.c: <— Received SIP response (557 bytes) from UDP:208.100.60.17:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bKPj1b6b58fd-28f1-4fed-9c6f-be332e97572b;received=192.168.0.14;rport=5060
From: sip:hiding_username@192.168.0.14:5060;tag=9318364c-ec29-4302-ab4d-f2266ef948be
To: sip:hiding_username@atlanta.voip.ms;tag=as2c9871bf
Call-ID: 8951857f-8080-47fe-a9e6-636c0cc4abbf
CSeq: 1646 OPTIONS
Server: voip.ms
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:208.100.60.17:5060
Accept: application/sdp
Content-Length: 0

Getting error: Everyone is busy/congested at this time
on voipms : Call Detail Records
I see the call but failed

finally i had just to pass a valid Caller ID
Thank you all for help

[sisco007] sisco007 https://community.asterisk.org/u/sisco007
February 25

getting error: Everyone is busy/congested at this time

Did you saw the 503 Service Unavailable error ?

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