[asterisk-users] congested/busy on trunk?
John Roman
john at dev1ce.com
Wed Mar 18 12:36:10 CDT 2020
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 <markster at digium.com>
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 at dunkel.dev1ce.com;transport=tcp SIP/2.0
Call-ID:
26b4820ac727b0f23fea131b7c5cd450 at 2605:e000:130a:fb:de1:71fc:e257:6f4e
CSeq: 8612 INVITE
From: "demo-alice"
<sip:demo-alice at dunkel.dev1ce.com;transport=tcp>;tag=3166828162
To: <sip:13107950860 at dunkel.dev1ce.com;transport=tcp>
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;branch=z9hG4bK00cb3358812774a5b9a5ea9aa42f8ffa373339;rport
Max-Forwards: 70
Contact: "demo-alice"
<sip:demo-alice@[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;transport=tcp>
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 --->
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 at 2605:e000:130a:fb:de1:71fc:e257:6f4e
From: "demo-alice" <sip:demo-alice at dunkel.dev1ce.com>;tag=3166828162
To:
<sip:13107950860 at dunkel.dev1ce.com>;tag=z9hG4bK00cb3358812774a5b9a5ea9aa42f8ffa373339
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 --->
ACK sip:13107950860 at dunkel.dev1ce.com;transport=tcp SIP/2.0
Call-ID:
26b4820ac727b0f23fea131b7c5cd450 at 2605:e000:130a:fb:de1:71fc:e257:6f4e
Max-Forwards: 70
From: "demo-alice"
<sip:demo-alice at dunkel.dev1ce.com;transport=tcp>;tag=3166828162
To:
<sip:13107950860 at dunkel.dev1ce.com>;tag=z9hG4bK00cb3358812774a5b9a5ea9aa42f8ffa373339
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 --->
INVITE sip:13107950860 at dunkel.dev1ce.com:5060;transport=tcp SIP/2.0
Call-ID:
26b4820ac727b0f23fea131b7c5cd450 at 2605:e000:130a:fb:de1:71fc:e257:6f4e
CSeq: 8613 INVITE
From: "demo-alice"
<sip:demo-alice at dunkel.dev1ce.com;transport=tcp>;tag=3166828162
To: <sip:13107950860 at dunkel.dev1ce.com;transport=tcp>
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;branch=z9hG4bKd67297ab9853f9168cf8c8533868ea6b373339;rport
Max-Forwards: 70
Contact: "demo-alice"
<sip:demo-alice@[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;transport=tcp>
Content-Type: application/sdp
Authorization: Digest
username="demo-alice",realm="dunkel.dev1ce.com",nonce="1584552767/f4c4bd9d5d9fb85b5292c7c5797b2c6a",uri="sip:13107950860 at 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'
<--- 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 at 2605:e000:130a:fb:de1:71fc:e257:6f4e
From: "demo-alice" <sip:demo-alice at dunkel.dev1ce.com>;tag=3166828162
To: <sip:13107950860 at dunkel.dev1ce.com>
CSeq: 8613 INVITE
Server: Asterisk PBX GIT-master-0cde95ec89
Content-Length: 0
-- Executing [13107950860 at anveo_sip:1]
Dial("PJSIP/demo-alice-00000002", "PJSIP/13107950860 at mytrunk") in
new stack
-- Called PJSIP/13107950860 at 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 at 2605:e000:130a:fb:de1:71fc:e257:6f4e
From: "demo-alice"
<sip:demo-alice at dunkel.dev1ce.com>;tag=3166828162
To:
<sip:13107950860 at dunkel.dev1ce.com>;tag=f1b212ab-9b55-4d13-9055-f49ce55f214e
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 --->
ACK sip:13107950860 at dunkel.dev1ce.com:5060;transport=tcp
SIP/2.0
Call-ID:
26b4820ac727b0f23fea131b7c5cd450 at 2605:e000:130a:fb:de1:71fc:e257:6f4e
Max-Forwards: 70
From: "demo-alice"
<sip:demo-alice at dunkel.dev1ce.com;transport=tcp>;tag=3166828162
To:
<sip:13107950860 at dunkel.dev1ce.com>;tag=f1b212ab-9b55-4d13-9055-f49ce55f214e
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;branch=z9hG4bKd67297ab9853f9168cf8c8533868ea6b373339;rport
CSeq: 8613 ACK
Content-Length: 0
On Tue, Mar 17, 2020 at 10:48:13AM -0300, Joshua C. Colp wrote:
> On Sat, Mar 14, 2020 at 2:02 PM John Roman <john at dev1ce.com> wrote:
>
> > 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 at anveo_sip:1] Dial("PJSIP/demo-alice-00000005",
> > "PJSIP/blah at mytrunk") in new stack
> > -- Called PJSIP/blah at 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 :)
> >
>
> 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.
>
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> _____________________________________________________________________
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>
> Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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--
-- -- --
john at dev1ce.com
https://dev1ce.com/john.gpg
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