Congested/busy On Trunk?
greetings asterisk users 🙂
ive just deployed version 17 and migrated as best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial:
Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890)
dunkel*CLI>
dunkel*CLI>
== Setting global variable ‘SIPDOMAIN’ to ‘ringythingy.dev1ce.com’
— Executing [blah@anveo_sip:1] Dial(“PJSIP/demo-alice-00000005”, “PJSIP/blah@mytrunk”) in new stack
— Called PJSIP/blah@mytrunk
— PJSIP/mytrunk-00000006 is ringing
— PJSIP/mytrunk-00000006 is ringing
— PJSIP/mytrunk-00000006 is making progress passing it to PJSIP/demo-alice-00000005
> 0x7ff39839e360 — Strict RTP learning after remote address set to: 72.9.156.128:52642
> 0x7ff3983994c0 — Strict RTP learning after remote address set to: [2605:e000:130a:fb:517d:7894:9482:c2bd]:54006
— PJSIP/mytrunk-00000006 is making progress passing it to PJSIP/demo-alice-00000005
== Everyone is busy/congested at this time (1:1/0/0)
— Auto fallthrough, channel ‘PJSIP/demo-alice-00000005’ status is ‘BUSY’
Any idea what im doing wrong? Thanks 🙂
—
— — —
john@dev1ce.com https://dev1ce.com/john.gpg
—
4 thoughts on - Congested/busy On Trunk?
The remote side eventually terminated the call. You’d need to grab a SIP
trace (pjsip set logger on) and provide/look at the actual traffic to see what is going on.
Based on your version string I also don’t believe you are on Asterisk 17, you appear to be on master which will become Asterisk 18.
ive enabled logging. aside from a realm error i see on my endpoint, im still not sure whats up
Asterisk GIT-master-0cde95ec89, Copyright (C) 1999 – 2018, Digium, Inc. and others. Created by Mark Spencer
;tag=3166828162
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’
for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type ‘core show license’ for details.
=========================================================================
Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel
(pid = 602055)
dunkel*CLI>
<--- Received SIP request (940 bytes) from TCP:[2605:e000:130a:fb:de1:71fc:e257:6f4e]:37879 --->
INVITE sip:13107950860@dunkel.dev1ce.com;transport=tcp SIP/2.0
Call-ID:
26b4820ac727b0f23fea131b7c5cd450@2605:e000:130a:fb:de1:71fc:e257:6f4e CSeq: 8612 INVITE
From: “demo-alice”
To:
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;branch=z9hG4bK00cb3358812774a5b9a5ea9aa42f8ffa373339;rport Max-Forwards: 70
Contact: “demo-alice”
Content-Type: application/sdp Content-Length: 345
v=0
o=- 1584552772838 1584552772841 IN IP6
2605:e000:130a:fb:de1:71fc:e257:6f4e s=-
c=IN IP6 2605:e000:130a:fb:de1:71fc:e257:6f4e t=0 0
m=audio 60954 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<--- Transmitting SIP response (681 bytes) to TCP:[2605:e000:130a:fb:de1:71fc:e257:6f4e]:37879 --->;tag=3166828162
;tag=z9hG4bK00cb3358812774a5b9a5ea9aa42f8ffa373339
SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;rport=37879;received=2605:e000:130a:fb:de1:71fc:e257:6f4e;branch=z9hG4bK00cb3358812774a5b9a5ea9aa42f8ffa373339
Call-ID:
26b4820ac727b0f23fea131b7c5cd450@2605:e000:130a:fb:de1:71fc:e257:6f4e From: “demo-alice”
To:
CSeq: 8612 INVITE
WWW-Authenticate: Digest realm=”dunkel.dev1ce.com”,nonce=”1584552767/f4c4bd9d5d9fb85b5292c7c5797b2c6a”,opaque=”733bad0a5366d9f2″,algorithm=md5,qop=”auth”
Server: Asterisk PBX GIT-master-0cde95ec89
Content-Length: 0
<--- Received SIP request (493 bytes) from TCP:[2605:e000:130a:fb:de1:71fc:e257:6f4e]:37879 --->
;tag=3166828162
;tag=z9hG4bK00cb3358812774a5b9a5ea9aa42f8ffa373339
ACK sip:13107950860@dunkel.dev1ce.com;transport=tcp SIP/2.0
Call-ID:
26b4820ac727b0f23fea131b7c5cd450@2605:e000:130a:fb:de1:71fc:e257:6f4e Max-Forwards: 70
From: “demo-alice”
To:
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;branch=z9hG4bK00cb3358812774a5b9a5ea9aa42f8ffa373339;rport CSeq: 8612 ACK
Content-Length: 0
<--- Received SIP request (1245 bytes) from TCP:[2605:e000:130a:fb:de1:71fc:e257:6f4e]:37879 --->
;tag=3166828162
INVITE sip:13107950860@dunkel.dev1ce.com:5060;transport=tcp SIP/2.0
Call-ID:
26b4820ac727b0f23fea131b7c5cd450@2605:e000:130a:fb:de1:71fc:e257:6f4e CSeq: 8613 INVITE
From: “demo-alice”
To:
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;branch=z9hG4bKd67297ab9853f9168cf8c8533868ea6b373339;rport Max-Forwards: 70
Contact: “demo-alice”
Content-Type: application/sdp Authorization: Digest username=”demo-alice”,realm=”dunkel.dev1ce.com”,nonce=”1584552767/f4c4bd9d5d9fb85b5292c7c5797b2c6a”,uri=”sip:13107950860@dunkel.dev1ce.com:5060;transport=tcp”,response=”6c4a62c6b4061e4b9312910a974abc4b”,algorithm=md5,opaque=”733bad0a5366d9f2″,qop=auth,cnonce=”xyz”,nc=00000001
Content-Length: 345
v=0
o=- 1584552772838 1584552772841 IN IP6
2605:e000:130a:fb:de1:71fc:e257:6f4e s=-
c=IN IP6 2605:e000:130a:fb:de1:71fc:e257:6f4e t=0 0
m=audio 60954 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
== Setting global variable ‘SIPDOMAIN’ to ‘dunkel.dev1ce.com’;tag=3166828162
<--- Transmitting SIP response (470 bytes) to TCP:[2605:e000:130a:fb:de1:71fc:e257:6f4e]:37879 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;rport=37879;received=2605:e000:130a:fb:de1:71fc:e257:6f4e;branch=z9hG4bKd67297ab9853f9168cf8c8533868ea6b373339
Call-ID:
26b4820ac727b0f23fea131b7c5cd450@2605:e000:130a:fb:de1:71fc:e257:6f4e
From: “demo-alice”
To:
CSeq: 8613 INVITE
Server: Asterisk PBX GIT-master-0cde95ec89
Content-Length: 0
— Executing [13107950860@anveo_sip:1]
;tag=3166828162
;tag=f1b212ab-9b55-4d13-9055-f49ce55f214e
Dial(“PJSIP/demo-alice-00000002”, “PJSIP/13107950860@mytrunk”) in
new stack
— Called PJSIP/13107950860@mytrunk
== Everyone is busy/congested at this time (1:0/1/0)
— Auto fallthrough, channel ‘PJSIP/demo-alice-00000002’
status is ‘CONGESTION’
<--- Transmitting SIP response (548 bytes) to TCP:[2605:e000:130a:fb:de1:71fc:e257:6f4e]:37879 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;rport=37879;received=2605:e000:130a:fb:de1:71fc:e257:6f4e;branch=z9hG4bKd67297ab9853f9168cf8c8533868ea6b373339
Call-ID:
26b4820ac727b0f23fea131b7c5cd450@2605:e000:130a:fb:de1:71fc:e257:6f4e
From: “demo-alice”
To:
CSeq: 8613 INVITE
Server: Asterisk PBX GIT-master-0cde95ec89
Reason: Q.850;cause=34
Content-Length: 0
<--- Received SIP request (489 bytes) from TCP:[2605:e000:130a:fb:de1:71fc:e257:6f4e]:37879 --->
;tag=3166828162
;tag=f1b212ab-9b55-4d13-9055-f49ce55f214e
ACK sip:13107950860@dunkel.dev1ce.com:5060;transport=tcp
SIP/2.0
Call-ID:
26b4820ac727b0f23fea131b7c5cd450@2605:e000:130a:fb:de1:71fc:e257:6f4e
Max-Forwards: 70
From: “demo-alice”
To:
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;branch=z9hG4bKd67297ab9853f9168cf8c8533868ea6b373339;rport
CSeq: 8613 ACK
Content-Length: 0
—
— — —
john@dev1ce.com https://dev1ce.com/john.gpg
—
Did you selectively enable logging? I don’t see any SIP request for the trunk. If you did enable it for everything, then I’d suggest checking
“pjsip show endpoints” and seeing the status of the trunk.
look for ‘mytrunk’ as thats the trunk its dialing
—
— — —
john@dev1ce.com https://dev1ce.com/john.gpg
—