Error: Everyone is busy/congested at this time

MAcomp_gd · September 21, 2020, 2:14pm

Hello everyone.
On a customer’s new PBX that we’re setting up with Asterisk 16, we get the following errors when trying to make an outbound call from an internal phone correctly registered to Asterisk to a mobile phone.
Unfortunately, Asterisk outputs the error “Everyone is busy/congested at this time”.

Here I provide the full log+pjsip trace of the attempt:

[Sep 21 09:41:58] VERBOSE[2473335][C-00000009] pbx.c: Executing [3484410XXX@internal:1] Dial("PJSIP/13-0000000c", "PJSIP/3484410XXX@BBBELL-endpoint,300,tT") in new stack
[Sep 21 09:41:58] VERBOSE[2473335][C-00000009] app_dial.c: Called PJSIP/3484410XXX@BBBELL-endpoint
[Sep 21 09:41:58] VERBOSE[2328342] res_pjsip_logger.c: <--- Transmitting SIP request (954 bytes) to UDP:192.168.1.6:5060 --->
INVITE sip:3484410XXX@sip.local.bbbell:5060 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bKPjaea26079-3f7f-499e-8d57-1efd583af88a^M
From: "BBG" <sip:13@sip.local.bbbell>;tag=0f3605cf-7c90-40e3-8542-740fec77a6c5^M
To: <sip:3484410XXX@sip.local.bbbell>^M
Contact: <sip:asterisk@192.168.1.5:5060>^M
Call-ID: 29270879-98ad-46d3-9007-503d58217625^M
CSeq: 29912 INVITE^M
Route: <sip:192.168.1.6;lr>^M
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER^M
Supported: 100rel, timer, replaces, norefersub^M
Session-Expires: 1800^M
Min-SE: 90^M
Max-Forwards: 70^M
User-Agent: Asterisk PBX certified/16.8-cert3^M
Content-Type: application/sdp^M
Content-Length:   235^M
^M
v=0^M
o=- 1543382357 1543382357 IN IP4 192.168.1.5^M
s=Asterisk^M
c=IN IP4 192.168.1.5^M
t=0 0^M
m=audio 20916 RTP/AVP 8 101^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=ptime:20^M
a=maxptime:150^M
a=sendrecv^M

[Sep 21 09:41:58] VERBOSE[2173954] res_pjsip_logger.c: <--- Received SIP response (439 bytes) from UDP:192.168.1.6:5060 --->
SIP/2.0 100 Trying^M
Via: SIP/2.0/UDP 192.168.1.5:5060;rport=5060;branch=z9hG4bKPjaea26079-3f7f-499e-8d57-1efd583af88a;received=192.168.1.5^M
From: "BBG" <sip:13@sip.local.bbbell>;tag=0f3605cf-7c90-40e3-8542-740fec77a6c5^M
To: <sip:3484410XXX@sip.local.bbbell>^M
Call-ID: 29270879-98ad-46d3-9007-503d58217625^M
CSeq: 29912 INVITE^M
Server: Patton SN5200 4B EUI 00A0BA1004BE R6.9 2017-03-13 H323 SIP M5T SIP Stack/4.2.14.18^M
Content-Length: 0^M
^M

[Sep 21 09:41:58] VERBOSE[2173954] res_pjsip_logger.c: <--- Received SIP response (522 bytes) from UDP:192.168.1.6:5060 --->
SIP/2.0 183 Session Progress^M
Via: SIP/2.0/UDP 192.168.1.5:5060;rport=5060;branch=z9hG4bKPjaea26079-3f7f-499e-8d57-1efd583af88a;received=192.168.1.5^M
From: "BBG" <sip:13@sip.local.bbbell>;tag=0f3605cf-7c90-40e3-8542-740fec77a6c5^M
To: <sip:3484410XXX@sip.local.bbbell>;tag=2444414513^M
Call-ID: 29270879-98ad-46d3-9007-503d58217625^M
CSeq: 29912 INVITE^M
Contact: <sip:3484410XXX@192.168.1.6:5060;transport=udp>^M
Server: Patton SN5200 4B EUI 00A0BA1004BE R6.9 2017-03-13 H323 SIP M5T SIP Stack/4.2.14.18^M
Content-Length: 0^M
^M

[Sep 21 09:41:58] VERBOSE[2473335][C-00000009] app_dial.c: PJSIP/BBBELL-endpoint-0000000d is making progress passing it to PJSIP/13-0000000c
[Sep 21 09:41:58] VERBOSE[2473335][C-00000009] app_dial.c: PJSIP/BBBELL-endpoint-0000000d is making progress passing it to PJSIP/13-0000000c
[Sep 21 09:41:58] VERBOSE[2328342] res_rtp_asterisk.c: 0x7f07e802af50 -- Strict RTP learning after remote address set to: 192.168.1.222:5004

[Sep 21 09:41:58] VERBOSE[2328342] res_pjsip_logger.c: <--- Transmitting SIP response (822 bytes) to UDP:192.168.1.222:5060 --->
SIP/2.0 183 Session Progress^M
Via: SIP/2.0/UDP 192.168.1.222:5060;rport=5060;received=192.168.1.222;branch=z9hG4bK525292603^M
Call-ID: 1146442903-5060-2@BJC.BGI.B.CCC^M
From: "BBG" <sip:13@192.168.1.5>;tag=1412627019^M
To: <sip:3484410XXX@192.168.1.5>;tag=fd5699f7-1f76-4fa0-8556-87cb50f4b991^M
CSeq: 11 INVITE^M
Server: Asterisk PBX certified/16.8-cert3^M
Contact: <sip:192.168.1.5:5060>^M
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER^M
Content-Type: application/sdp^M
Content-Length:   271^M
^M
v=0^M
o=- 8000 8002 IN IP4 192.168.1.5^M
s=Asterisk^M
c=IN IP4 192.168.1.5^M
t=0 0^M
m=audio 17660 RTP/AVP 9 0 8 101^M
a=rtpmap:9 G722/8000^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=ptime:20^M
a=maxptime:150^M
a=sendrecv^M

[Sep 21 09:41:58] VERBOSE[2328342] res_pjsip_logger.c: <--- Transmitting SIP response (822 bytes) to UDP:192.168.1.222:5060 --->
SIP/2.0 183 Session Progress^M
Via: SIP/2.0/UDP 192.168.1.222:5060;rport=5060;received=192.168.1.222;branch=z9hG4bK525292603^M
Call-ID: 1146442903-5060-2@BJC.BGI.B.CCC^M
From: "BBG" <sip:13@192.168.1.5>;tag=1412627019^M
To: <sip:3484410XXX@192.168.1.5>;tag=fd5699f7-1f76-4fa0-8556-87cb50f4b991^M
CSeq: 11 INVITE^M
Server: Asterisk PBX certified/16.8-cert3^M
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER^M
Contact: <sip:192.168.1.5:5060>^M
Content-Type: application/sdp^M
Content-Length:   271^M
^M
v=0^M
o=- 8000 8002 IN IP4 192.168.1.5^M
s=Asterisk^M
c=IN IP4 192.168.1.5^M
t=0 0^M
m=audio 17660 RTP/AVP 9 0 8 101^M
a=rtpmap:9 G722/8000^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=ptime:20^M
a=maxptime:150^M
a=sendrecv^M

[Sep 21 09:41:59] VERBOSE[2473335][C-00000009] res_rtp_asterisk.c: 0x7f07e802af50 -- Strict RTP switching to RTP target address 192.168.1.222:5004 as source
[Sep 21 09:41:59] VERBOSE[2173954] res_pjsip_logger.c: <--- Received SIP response (502 bytes) from UDP:192.168.1.6:5060 --->
SIP/2.0 403 Forbidden^M
Via: SIP/2.0/UDP 192.168.1.5:5060;rport=5060;branch=z9hG4bKPjaea26079-3f7f-499e-8d57-1efd583af88a;received=192.168.1.5^M
From: "BBG" <sip:13@sip.local.bbbell>;tag=0f3605cf-7c90-40e3-8542-740fec77a6c5^M
To: <sip:3484410XXX@sip.local.bbbell>;tag=2444414513^M
Call-ID: 29270879-98ad-46d3-9007-503d58217625^M
CSeq: 29912 INVITE^M
Server: Patton SN5200 4B EUI 00A0BA1004BE R6.9 2017-03-13 H323 SIP M5T SIP Stack/4.2.14.18^M
Reason: Q.850;cause=21;text="Call rejected"^M
Content-Length: 0^M
^M

[Sep 21 09:41:59] VERBOSE[2328342] res_pjsip_logger.c: <--- Transmitting SIP request (458 bytes) to UDP:192.168.1.6:5060 --->
ACK sip:3484410XXX@sip.local.bbbell:5060 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bKPjaea26079-3f7f-499e-8d57-1efd583af88a^M
From: "BBG" <sip:13@sip.local.bbbell>;tag=0f3605cf-7c90-40e3-8542-740fec77a6c5^M
To: <sip:3484410XXX@sip.local.bbbell>;tag=2444414513^M
Call-ID: 29270879-98ad-46d3-9007-503d58217625^M
CSeq: 29912 ACK^M
Route: <sip:192.168.1.6;lr>^M
Max-Forwards: 70^M
User-Agent: Asterisk PBX certified/16.8-cert3^M
Content-Length:  0^M
^M

[Sep 21 09:41:59] VERBOSE[2473335][C-00000009] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[Sep 21 09:41:59] VERBOSE[2473335][C-00000009] pbx.c: Executing [3484410XXX@internal:2] Congestion("PJSIP/13-0000000c", "") in new stack

[Sep 21 09:41:59] VERBOSE[2473336] res_pjsip_logger.c: <--- Transmitting SIP response (511 bytes) to UDP:192.168.1.222:5060 --->
SIP/2.0 503 Service Unavailable^M
Via: SIP/2.0/UDP 192.168.1.222:5060;rport=5060;received=192.168.1.222;branch=z9hG4bK525292603^M
Call-ID: 1146442903-5060-2@BJC.BGI.B.CCC^M
From: "BBG" <sip:13@192.168.1.5>;tag=1412627019^M
To: <sip:3484410XXX@192.168.1.5>;tag=fd5699f7-1f76-4fa0-8556-87cb50f4b991^M
CSeq: 11 INVITE^M
Server: Asterisk PBX certified/16.8-cert3^M
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER^M
Reason: Q.850;cause=34^M
Content-Length:  0^M
^M

[Sep 21 09:41:59] VERBOSE[2473335][C-00000009] pbx.c: Spawn extension (internal, 3484410XXX, 2) exited non-zero on 'PJSIP/13-0000000c'
[Sep 21 09:41:59] VERBOSE[2173954] res_pjsip_logger.c: <--- Received SIP request (308 bytes) from UDP:192.168.1.222:5060 --->
ACK sip:3484410XXX@192.168.1.5 SIP/2.0^M
Via: SIP/2.0/UDP 192.168.1.222:5060;branch=z9hG4bK525292603;rport^M
From: "BBG" <sip:13@192.168.1.5>;tag=1412627019^M
To: <sip:3484410XXX@192.168.1.5>;tag=fd5699f7-1f76-4fa0-8556-87cb50f4b991^M
Call-ID: 1146442903-5060-2@BJC.BGI.B.CCC^M
CSeq: 11 ACK^M
Content-Length: 0^M
^M

Can someone kindly help me to troubleshoot this error ? I suspect it might be something on the outbound proxy given the 403 error in the pjsip trace, but i have no control over what the proxy is doing. I’d like to make sure it’s not something Asterisk related before contacting my ISP.
Many thanks.

david551 · September 21, 2020, 2:23pm

The called party is saying that you don’t have permission to make the call. That could be that they don’t recognize you as a valid user, or could be that you are trying to call a number that is not permitted.

As there is no request for authentication data, I tend to suspect the former.

Please note that this forum is not appropriate if you want answers in terms of the GUI forms.

MAcomp_gd · September 22, 2020, 8:20am

The number I’m trying to call is just a regular national mobile phone number and by the contract I made with the ISP I should be able to make the call.

Just now the ISP told me that in the field:

From: "BBG" <sip:13@sip.local.bbbell>

They expected the username of the account that is making the call, but the username is in fact “BBG”, so I’m confused. The phone is correctly registered to Asterisk, the endpoint exists and it’s available and as far as I know its AoR is properly set up.
Should I include my pjsip.conf then ?

Also, what do you mean by “Please note that this forum is not appropriate if you want answers in terms of the GUI forms.” ?

MAcomp_gd · September 22, 2020, 8:50am

Well, turns out I didn’t understand what they told me. I think now they intended that the call should figure with the username used to register the sip trunk and not directly with the username of the endpoint that is making the call.
But how can I do such a thing ?

david551 · September 22, 2020, 10:15am

;from_user=     ; Username to use in From header for requests to this endpoint
                ; (default: "")

The reference to GUIs meant that any answer we give will be in terms of what goes in pjsip.conf, extensions.conf, etc., not in terms of what you enter on a FreePBX form. As this is a common requirement, there will be a way of doing it in FreePBX, but not everything that can be done with Asterisk can be done with FreePBX.

system · October 22, 2020, 10:15am

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

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