Everyone is busy/congested at this time OUTBOUND Call error

jaydeepkasture · January 23, 2024, 7:44am

I have two Asterisk servers, labeled as 001 and 002. Each server hosts a set of users.

On Asterisk 001, the users are as follows:

User 001, a WebRTC user.
Users 100 to 105, all using UDP.

Similarly, on Asterisk 002, the setup includes:

User 002, a WebRTC user.
Users 200 to 205, all using UDP.

The configuration allows users from Asterisk server 001 to call users on Asterisk server 002. Calls between UDP users, using softphones like Zoiper and MicroSIP, are successful. However, issues arise with WebRTC users (001 and 002). When they attempt calls using softphones, an error stating “Everyone is busy/congested at this time” is encountered. Interestingly, these calls work smoothly when using the Browser Phone solution from Innovate Asterisk’s GitHub.

Additionally, the same error occurs when a UDP user on a softphone tries to call a WebRTC user who is also using a softphone.

Asterisk 001 PJSIP


[global]
type=global
user_agent=Asterisk-001

[transport-wss]
type=transport
protocol=wss ; Use 'wss' for WebSocket Secure
bind=0.0.0.0 ; Replace with your Asterisk server's IP and secure port

[udp]
type=transport
protocol=udp
bind=0.0.0.0
tos=af42
cos=3

[tcp]
type=transport
protocol=tcp
bind=0.0.0.0

[endpoint-basic](!)
type=endpoint
transport=transport-wss
context=outside ; Replace with the context you use in your dialplan
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
auth=auth_user
aors=sip_user
webrtc=yes ; Enable WebRTC support
direct_media = no
rtp_timeout=30

; Authentication for the User
[auth_user](!)
type=auth
auth_type=userpass

; AOR for the User
[sip_user](!)
type=aor
max_contacts=5 ; Adjust as needed

; Specific User Configuration
[001](endpoint-basic)
auth=auth001
aors=001

[auth001](auth_user)
username=001 ; Replace with your username
password=pwd; Replace with a strong password

[001](sip_user)

[User2](endpoint-basic)
auth=authUser2
aors=User2

[authUser2](auth_user)
username=User2 ; Replace with your username
password=pwd; Replace with a strong password

[User2](sip_user)

;==================softphone=============
[auth100]
type=auth
username=100
password=pwd
auth_type=userpass

[100]
type=aor
max_contacts=2
qualify_frequency=60

[100]
type=endpoint
context=outside
auth=auth100
aors=100
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media = no

[auth101]
type=auth
username=101
password=pwd
auth_type=userpass

[101]
type=aor
max_contacts=2
qualify_frequency=60

[101]
type=endpoint
context=outside
auth=auth101
aors=101
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media=no

[auth102]
type=auth
username=102
password=pwd
auth_type=userpass

[102]
type=aor
max_contacts=2
qualify_frequency=60

[102]
type=endpoint
context=outside
auth=auth102
aors=102
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media=no

[auth103]
type=auth
username=103
password=pwd
auth_type=userpass

[103]
type=aor
max_contacts=2
qualify_frequency=60

[103]
type=endpoint
context=outside
auth=auth103
aors=103
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media=no

[auth104]
type=auth
username=104
password=pwd
auth_type=userpass

[104]
type=aor
max_contacts=2
qualify_frequency=60

[104]
type=endpoint
context=outside
auth=auth104
aors=104
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media=no

[auth105]
type=auth
username=105
password=pwd
auth_type=userpass

[105]
type=aor
max_contacts=2
qualify_frequency=60

[105]
type=endpoint
context=outside
auth=auth105
aors=105
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media=no


;=========002====


[Asterisk-002]
type=aor
contact=sip:ip-address

[Asterisk-002]
type=endpoint
context=from-002
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
aors=002
direct_media = no

[Asterisk-002]
type=identify
endpoint=002
match=ip-address

Asterisk 002 PJSIP


[global]
type=global
user_agent=002

[transport-wss]
type=transport
protocol=wss ; Use 'wss' for WebSocket Secure
bind=0.0.0.0 ; Replace with your Asterisk server's IP and secure port

[udp]
type=transport
protocol=udp
bind=0.0.0.0
tos=af42
cos=3

[tcp]
type=transport
protocol=tcp
bind=0.0.0.0

[endpoint-basic](!)
type=endpoint
transport=transport-wss
context=outside ; Replace with the context you use in your dialplan
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
auth=auth_user
aors=sip_user
webrtc=yes ; Enable WebRTC support
direct_media = no
rtp_timeout=30

; Authentication for the User
[auth_user](!)
type=auth
auth_type=userpass

; AOR for the User
[sip_user](!)
type=aor
max_contacts=5 ; Adjust as needed

; Specific User Configuration
[002](endpoint-basic)
auth=auth002
aors=002

[auth002](auth_user)
username=002 ; Replace with your username
password=pwd; Replace with a strong password

[002](sip_user)

[User2](endpoint-basic)
auth=authUser2
aors=User2

[authUser2](auth_user)
username=User2 ; Replace with your username
password=pwd; Replace with a strong password

[User2](sip_user)

;==================softphone=============
[auth200]
type=auth
username=200
password=pwd
auth_type=userpass

[200]
type=aor
max_contacts=2
qualify_frequency=60

[200]
type=endpoint
context=outside
auth=auth200
aors=200
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media = no

[auth201]
type=auth
username=201
password=pwd
auth_type=userpass

[201]
type=aor
max_contacts=2
qualify_frequency=60

[201]
type=endpoint
context=outside
auth=auth201
aors=201
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media=no

[auth202]
type=auth
username=202
password=pwd
auth_type=userpass

[202]
type=aor
max_contacts=2
qualify_frequency=60

[202]
type=endpoint
context=outside
auth=auth202
aors=202
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media=no

[auth203]
type=auth
username=203
password=pwd
auth_type=userpass

[203]
type=aor
max_contacts=2
qualify_frequency=60

[203]
type=endpoint
context=outside
auth=auth203
aors=203
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media=no

[auth204]
type=auth
username=204
password=pwd
auth_type=userpass

[204]
type=aor
max_contacts=2
qualify_frequency=60

[204]
type=endpoint
context=outside
auth=auth204
aors=204
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media=no

[auth205]
type=auth
username=205
password=pwd
auth_type=userpass

[205]
type=aor
max_contacts=2
qualify_frequency=60

[205]
type=endpoint
context=outside
auth=auth205
aors=205
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
direct_media=no


;=========002====


[Asterisk-001]
type=aor
contact=sip:ip-address

[Asterisk-001]
type=endpoint
context=from-001
transport=tcp
disallow=all
allow=opus,ulaw,alaw,vp8,vp9
aors=Asterisk-001
direct_media = no

[Asterisk-001]
type=identify
endpoint=001
match=ip-address

jcolp · January 23, 2024, 9:26am

You should actually provide Asterisk logging and SIP traces (pjsip set logger on).

jaydeepkasture · January 24, 2024, 4:38pm

Senders Log

<--- Received SIP request (1001 bytes) from UDP:192.168.1.9:34125 --->
INVITE sip:002@192.168.1.9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:34125;branch=z9hG4bK-524287-1---61b1846e569241dd;rport
Max-Forwards: 70
Contact: <sip:001@192.168.1.9:34125;transport=UDP>
To: <sip:002@192.168.1.9>
From: <sip:001@192.168.1.9;transport=UDP>;tag=75eb506f
Call-ID: DpxcCyZCNOAGB9_ou_WyFg..
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: 336

v=0
o=Z 0 690429 IN IP4 192.168.1.9
s=Z
c=IN IP4 192.168.1.9
t=0 0
m=audio 47994 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 (481 bytes) to UDP:192.168.1.9:34125 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.9:34125;rport=34125;received=192.168.1.9;branch=z9hG4bK-524287-1---61b1846e569241dd
Call-ID: DpxcCyZCNOAGB9_ou_WyFg..
From: <sip:001@192.168.1.9>;tag=75eb506f
To: <sip:002@192.168.1.9>;tag=z9hG4bK-524287-1---61b1846e569241dd
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1706114483/5cb18b83e14c4ed471f28d45f7210783",opaque="78b93dfc64ce6b34",algorithm=MD5,qop="auth"
Server: ULG 1
Content-Length:  0


<--- Received SIP request (342 bytes) from UDP:192.168.1.9:34125 --->
ACK sip:002@192.168.1.9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:34125;branch=z9hG4bK-524287-1---61b1846e569241dd;rport
Max-Forwards: 70
To: <sip:002@192.168.1.9>;tag=z9hG4bK-524287-1---61b1846e569241dd
From: <sip:001@192.168.1.9;transport=UDP>;tag=75eb506f
Call-ID: DpxcCyZCNOAGB9_ou_WyFg..
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1295 bytes) from UDP:192.168.1.9:34125 --->
INVITE sip:002@192.168.1.9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:34125;branch=z9hG4bK-524287-1---5393da6099d30f36;rport
Max-Forwards: 70
Contact: <sip:001@192.168.1.9:34125;transport=UDP>
To: <sip:002@192.168.1.9>
From: <sip:001@192.168.1.9;transport=UDP>;tag=75eb506f
Call-ID: DpxcCyZCNOAGB9_ou_WyFg..
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="001",realm="asterisk",nonce="1706114483/5cb18b83e14c4ed471f28d45f7210783",uri="sip:002@192.168.1.9;transport=UDP",response="ab3e73d03eedc747249340ef1073b618",cnonce="9005384791b79506752986f42a2ac75f",nc=00000001,qop=auth,algorithm=MD5,opaque="78b93dfc64ce6b34"
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 336

v=0
o=Z 0 690429 IN IP4 192.168.1.9
s=Z
c=IN IP4 192.168.1.9
t=0 0
m=audio 47994 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 (289 bytes) to UDP:192.168.1.9:34125 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.9:34125;rport=34125;received=192.168.1.9;branch=z9hG4bK-524287-1---5393da6099d30f36
Call-ID: DpxcCyZCNOAGB9_ou_WyFg..
From: <sip:001@192.168.1.9>;tag=75eb506f
To: <sip:002@192.168.1.9>
CSeq: 2 INVITE
Server: ULG 1
Content-Length:  0


    -- Executing [002@outside:1] NoOp("PJSIP/001-00000008", "Calling: 002)") in new stack
    -- Executing [002@outside:2] Dial("PJSIP/001-00000008", "PJSIP/002@outside-call") in new stack
<--- Transmitting SIP request (1052 bytes) to UDP:192.168.255.134:5060 --->
INVITE sip:002@192.168.255.134 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.131:5060;rport;branch=z9hG4bKPj7f44b253-6c94-4c2c-afa2-7ade030d2971
From: <sip:001@192.168.255.131>;tag=c864970c-0ac1-4175-9fa6-9a2795dea4ef
To: <sip:002@192.168.255.134>
Contact: <sip:asterisk@192.168.255.131:5060>
Call-ID: e74481a0-147d-4ec5-95c1-2a5ca71fbbe4
CSeq: 13694 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:001@192.168.255.131>
Max-Forwards: 70
User-Agent: ULG 1
Content-Type: application/sdp
Content-Length:   351

v=0
o=- 2137557473 2137557473 IN IP4 192.168.255.131
s=Asterisk
c=IN IP4 192.168.255.131
t=0 0
m=audio 10954 RTP/AVP 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

    -- Called PJSIP/002@outside-call
<--- Received SIP response (357 bytes) from UDP:192.168.255.134:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.255.131:5060;rport=5060;received=192.168.255.131;branch=z9hG4bKPj7f44b253-6c94-4c2c-afa2-7ade030d2971
Call-ID: e74481a0-147d-4ec5-95c1-2a5ca71fbbe4
From: <sip:001@192.168.255.131>;tag=c864970c-0ac1-4175-9fa6-9a2795dea4ef
To: <sip:002@192.168.255.134>
CSeq: 13694 INVITE
Server: ULG 1
Content-Length:  0


<--- Received SIP response (482 bytes) from UDP:192.168.255.134:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.255.131:5060;rport=5060;received=192.168.255.131;branch=z9hG4bKPj7f44b253-6c94-4c2c-afa2-7ade030d2971
Call-ID: e74481a0-147d-4ec5-95c1-2a5ca71fbbe4
From: <sip:001@192.168.255.131>;tag=c864970c-0ac1-4175-9fa6-9a2795dea4ef
To: <sip:002@192.168.255.134>;tag=80a53a0d-8632-4d90-9e51-7e45fc9146cf
CSeq: 13694 INVITE
Server: ULG 1
Reason: Q.850;cause=28
P-Asserted-Identity: <sip:002@192.168.255.134>
Content-Length:  0


<--- Transmitting SIP request (404 bytes) to UDP:192.168.255.134:5060 --->
ACK sip:002@192.168.255.134 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.131:5060;rport;branch=z9hG4bKPj7f44b253-6c94-4c2c-afa2-7ade030d2971
From: <sip:001@192.168.255.131>;tag=c864970c-0ac1-4175-9fa6-9a2795dea4ef
To: <sip:002@192.168.255.134>;tag=80a53a0d-8632-4d90-9e51-7e45fc9146cf
Call-ID: e74481a0-147d-4ec5-95c1-2a5ca71fbbe4
CSeq: 13694 ACK
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [002@outside:3] Hangup("PJSIP/001-00000008", "") in new stack
  == Spawn extension (outside, 002, 3) exited non-zero on 'PJSIP/001-00000008'
<--- Transmitting SIP response (410 bytes) to UDP:192.168.1.9:34125 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.1.9:34125;rport=34125;received=192.168.1.9;branch=z9hG4bK-524287-1---5393da6099d30f36
Call-ID: DpxcCyZCNOAGB9_ou_WyFg..
From: <sip:001@192.168.1.9>;tag=75eb506f
To: <sip:002@192.168.1.9>;tag=548ba850-124c-4a3a-90f6-70bf0dbc10d8
CSeq: 2 INVITE
Server: ULG 1
Reason: Q.850;cause=28
P-Asserted-Identity: <sip:002@192.168.1.9>
Content-Length:  0


<--- Received SIP request (343 bytes) from UDP:192.168.1.9:34125 --->
ACK sip:002@192.168.1.9;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:34125;branch=z9hG4bK-524287-1---5393da6099d30f36;rport
Max-Forwards: 70
To: <sip:002@192.168.1.9>;tag=548ba850-124c-4a3a-90f6-70bf0dbc10d8
From: <sip:001@192.168.1.9;transport=UDP>;tag=75eb506f
Call-ID: DpxcCyZCNOAGB9_ou_WyFg..
CSeq: 2 ACK
Content-Length: 0

Receiver Logs

<--- Received SIP request (1052 bytes) from UDP:192.168.255.131:5060 --->
INVITE sip:002@192.168.255.134 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.131:5060;rport;branch=z9hG4bKPj7f44b253-6c94-4c2c-afa2-7ade030d2971
From: <sip:001@192.168.255.131>;tag=c864970c-0ac1-4175-9fa6-9a2795dea4ef
To: <sip:002@192.168.255.134>
Contact: <sip:asterisk@192.168.255.131:5060>
Call-ID: e74481a0-147d-4ec5-95c1-2a5ca71fbbe4
CSeq: 13694 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:001@192.168.255.131>
Max-Forwards: 70
User-Agent: ULG 1
Content-Type: application/sdp
Content-Length:   351

v=0
o=- 2137557473 2137557473 IN IP4 192.168.255.131
s=Asterisk
c=IN IP4 192.168.255.131
t=0 0
m=audio 10954 RTP/AVP 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Transmitting SIP response (357 bytes) to UDP:192.168.255.131:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.255.131:5060;rport=5060;received=192.168.255.131;branch=z9hG4bKPj7f44b253-6c94-4c2c-afa2-7ade030d2971
Call-ID: e74481a0-147d-4ec5-95c1-2a5ca71fbbe4
From: <sip:001@192.168.255.131>;tag=c864970c-0ac1-4175-9fa6-9a2795dea4ef
To: <sip:002@192.168.255.134>
CSeq: 13694 INVITE
Server: ULG 1
Content-Length:  0


    -- Executing [002@from-outside-call:1] NoOp("PJSIP/outside-call-00000016", "Calling: 002)") in new stack
    -- Executing [002@from-outside-call:2] Dial("PJSIP/outside-call-00000016", "PJSIP/002") in new stack
    -- Called PJSIP/002
<--- Transmitting SIP request (1712 bytes) to TCP:192.168.255.134:44275 --->
INVITE sip:002@192.168.255.134:44275;transport=TCP;rinstance=21040e6a5c98aa7f SIP/2.0
Via: SIP/2.0/TCP 192.168.255.134:5060;rport;branch=z9hG4bKPj46781e9a-10e3-496e-a9b2-0f09e3d55a41;alias
From: <sip:001@192.168.255.134>;tag=2bb0f9e5-cca2-4763-bd2b-b51901a4cc3c
To: <sip:002@192.168.255.134;rinstance=21040e6a5c98aa7f>
Contact: <sip:asterisk@192.168.255.134:5060;transport=TCP>
Call-ID: 1fd7792b-4288-40c8-8d19-72fce0040f6b
CSeq: 13486 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: <sip:001@192.168.255.134>
Max-Forwards: 70
User-Agent: ULG 1
Content-Type: application/sdp
Content-Length:   917

v=0
o=- 992855178 992855178 IN IP4 192.168.255.134
s=Asterisk
c=IN IP4 192.168.255.134
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0
m=audio 14398 UDP/TLS/RTP/SAVPF 0 8 107 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 E3:53:4C:6C:D9:0D:2A:E6:E3:AE:53:C4:E0:B6:1A:27:76:61:89:47:EE:CC:61:0B:A1:81:D1:84:D3:68:B2:96
a=ice-ufrag:54b742c8086f7a856509dabe2bceb72b
a=ice-pwd:57bbe0515e80f8291815c3cc193d5a05
a=candidate:Hc0a8ff86 1 UDP 2130706431 192.168.255.134 14398 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:328836633 cname:96b686b6-6b09-45eb-bc73-1c3c6d235d1c
a=msid:f9bc3e1d-819d-4787-8839-a68ade099fa3 617eb8a0-9bb9-4630-9ad1-2b9554c19cc2
a=rtcp-fb:* transport-cc
a=mid:audio-0

<--- Received SIP response (406 bytes) from TCP:192.168.255.134:44275 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/TCP 192.168.255.134:5060;rport=5060;branch=z9hG4bKPj46781e9a-10e3-496e-a9b2-0f09e3d55a41;alias
To: <sip:002@192.168.255.134;rinstance=21040e6a5c98aa7f>;tag=3b67cd66
From: <sip:001@192.168.255.134>;tag=2bb0f9e5-cca2-4763-bd2b-b51901a4cc3c
Call-ID: 1fd7792b-4288-40c8-8d19-72fce0040f6b
CSeq: 13486 INVITE
User-Agent: Z 5.6.2 v2.10.20.2
Content-Length: 0


<--- Transmitting SIP request (456 bytes) to TCP:192.168.255.134:44275 --->
ACK sip:002@192.168.255.134:44275;transport=TCP;rinstance=21040e6a5c98aa7f SIP/2.0
Via: SIP/2.0/TCP 192.168.255.134:5060;rport;branch=z9hG4bKPj46781e9a-10e3-496e-a9b2-0f09e3d55a41;alias
From: <sip:001@192.168.255.134>;tag=2bb0f9e5-cca2-4763-bd2b-b51901a4cc3c
To: <sip:002@192.168.255.134;rinstance=21040e6a5c98aa7f>;tag=3b67cd66
Call-ID: 1fd7792b-4288-40c8-8d19-72fce0040f6b
CSeq: 13486 ACK
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [002@from-outside-call:3] Hangup("PJSIP/outside-call-00000016", "") in new stack
  == Spawn extension (from-outside-call, 002, 3) exited non-zero on 'PJSIP/outside-call-00000016'
<--- Transmitting SIP response (482 bytes) to UDP:192.168.255.131:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 192.168.255.131:5060;rport=5060;received=192.168.255.131;branch=z9hG4bKPj7f44b253-6c94-4c2c-afa2-7ade030d2971
Call-ID: e74481a0-147d-4ec5-95c1-2a5ca71fbbe4
From: <sip:001@192.168.255.131>;tag=c864970c-0ac1-4175-9fa6-9a2795dea4ef
To: <sip:002@192.168.255.134>;tag=80a53a0d-8632-4d90-9e51-7e45fc9146cf
CSeq: 13694 INVITE
Server: ULG 1
Reason: Q.850;cause=28
P-Asserted-Identity: <sip:002@192.168.255.134>
Content-Length:  0


<--- Received SIP request (404 bytes) from UDP:192.168.255.131:5060 --->
ACK sip:002@192.168.255.134 SIP/2.0
Via: SIP/2.0/UDP 192.168.255.131:5060;rport;branch=z9hG4bKPj7f44b253-6c94-4c2c-afa2-7ade030d2971
From: <sip:001@192.168.255.131>;tag=c864970c-0ac1-4175-9fa6-9a2795dea4ef
To: <sip:002@192.168.255.134>;tag=80a53a0d-8632-4d90-9e51-7e45fc9146cf
Call-ID: e74481a0-147d-4ec5-95c1-2a5ca71fbbe4
CSeq: 13694 ACK
Max-Forwards: 70
User-Agent: ULG 1
Content-Length:  0


<--- Received SIP request (747 bytes) from TCP:192.168.255.1:60528 --->
REGISTER sip:192.168.255.134;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.255.1:60528;rport;branch=z9hG4bKPjb079bddcd3654da1a9ae9797d1359073;alias
Route: <sip:192.168.255.134;transport=tcp;lr>
Max-Forwards: 70
From: "microsip" <sip:1111@192.168.255.134>;tag=4a2081359b3642819915c1943e98a8e9
To: "microsip" <sip:1111@192.168.255.134>
Call-ID: 16665adedd4b4b4b8db6e38e5fa4624f
CSeq: 23674 REGISTER
User-Agent: MicroSIP/3.21.3
Supported: outbound, path
Contact: "microsip" <sip:1111@192.168.255.1:60528;transport=TCP;ob>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000039fb77c1>"
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


<--- Transmitting SIP response (573 bytes) to TCP:192.168.255.1:60528 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 192.168.255.1:60528;rport=60528;received=192.168.255.1;branch=z9hG4bKPjb079bddcd3654da1a9ae9797d1359073;alias
Call-ID: 16665adedd4b4b4b8db6e38e5fa4624f
From: "microsip" <sip:1111@192.168.255.134>;tag=4a2081359b3642819915c1943e98a8e9
To: "microsip" <sip:1111@192.168.255.134>;tag=z9hG4bKPjb079bddcd3654da1a9ae9797d1359073
CSeq: 23674 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1706114496/5f5d9f9912e559bdc095b9b161d52720",opaque="60d0213a2f9c524a",algorithm=MD5,qop="auth"
Server: ULG 1
Content-Length:  0


<--- Received SIP request (1051 bytes) from TCP:192.168.255.1:60528 --->
REGISTER sip:192.168.255.134;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.255.1:60528;rport;branch=z9hG4bKPjb4ebf8f709d5411aa2e6604b1256d3b0;alias
Route: <sip:192.168.255.134;transport=tcp;lr>
Max-Forwards: 70
From: "microsip" <sip:1111@192.168.255.134>;tag=4a2081359b3642819915c1943e98a8e9
To: "microsip" <sip:1111@192.168.255.134>
Call-ID: 16665adedd4b4b4b8db6e38e5fa4624f
CSeq: 23675 REGISTER
User-Agent: MicroSIP/3.21.3
Supported: outbound, path
Contact: "microsip" <sip:1111@192.168.255.1:60528;transport=TCP;ob>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000039fb77c1>"
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="1111", realm="asterisk", nonce="1706114496/5f5d9f9912e559bdc095b9b161d52720", uri="sip:192.168.255.134;transport=tcp", response="6fdb810c4b3ebfca3b145b01a328015e", algorithm=MD5, cnonce="e59ecd68c61e45d7bfb3a95ca494aa40", opaque="60d0213a2f9c524a", qop=auth, nc=00000001
Content-Length:  0


<--- Transmitting SIP response (538 bytes) to TCP:192.168.255.1:60528 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.255.1:60528;rport=60528;received=192.168.255.1;branch=z9hG4bKPjb4ebf8f709d5411aa2e6604b1256d3b0;alias
Call-ID: 16665adedd4b4b4b8db6e38e5fa4624f
From: "microsip" <sip:1111@192.168.255.134>;tag=4a2081359b3642819915c1943e98a8e9
To: "microsip" <sip:1111@192.168.255.134>;tag=z9hG4bKPjb4ebf8f709d5411aa2e6604b1256d3b0
CSeq: 23675 REGISTER
Date: Wed, 24 Jan 2024 16:41:36 GMT
Contact: <sip:1111@192.168.255.1:60528;transport=TCP;ob>;expires=299
Expires: 300
Server: ULG 1
Content-Length:  0


    -- Removed contact 'sip:001@192.168.255.134:44275;transport=TCP;rinstance=cf6fd097d757f0f4' from AOR '001' due to expiration

david551 · January 24, 2024, 4:56pm

Is an incomplete number.

This is a dialplan problem, but you have not provided your dialplan.

jaydeepkasture · January 25, 2024, 7:46am

The dialplan is quite simple both sender and receiver has the exact same dialplan


[outside]
exten => _XXX,1,NoOp(Calling: ${EXTEN}))
same => n,Dial(PJSIP/${EXTEN}@outside-call)
same => n,Hangup()



[from-outside-call]
exten => _XXX,1,NoOp(Calling: ${EXTEN}))
same => n,Dial(PJSIP/${EXTEN})
same => n,Hangup()

this dialplan is working if I use udp user in the receiver end and call from webrtc user using softphone

david551 · January 25, 2024, 11:03am

This is not the user agent you configured, and not a default Asterisk one.

jaydeepkasture · January 25, 2024, 3:34pm

I am new to asterisk I didn’t understand what you mean by This is not the user agent you configured

david551 · January 25, 2024, 5:28pm

This is what you configured.

I’m suggesting that whatever is responding isn’t Asterisk.

jaydeepkasture · January 26, 2024, 6:48am

Asterisk’s response is “Everyone is busy/congested at this time”
However both the user were online and not busy

jcolp · January 26, 2024, 9:54am

@jaydeepkasture Please stop creating a new post for this same problem.

jaydeepkasture · January 26, 2024, 10:02am

Sure thanks for the suggestion

david551 · January 26, 2024, 1:02pm

Everyone is busy… is a secondary error. As I pointed out the primary error is that you haven’t provided a long enough number. I was a little confused about what was reporting that. It is actually 192.168.255.134, which is the device with the user agent that I quoted, above.

jaydeepkasture · January 26, 2024, 4:21pm

Thanks for your reply @david551
I understood what you saying

user_agent=Asterisk-001
this was just the server name that I configured in the pjsip file in the globe section

system · February 25, 2024, 4:21pm

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