Re: [asterisk-users] congested/busy on trunk?
look for 'mytrunk' as thats the trunk its dialing On Wed, Mar 18, 2020 at 02:41:51PM -0300, Joshua C. Colp wrote: > On Wed, Mar 18, 2020 at 2:37 PM John Roman wrote: > > > ive enabled logging. aside from a realm error i see on my endpoint, im > > still not sure whats up > > > 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. > > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- -- -- -- j...@dev1ce.com https://dev1ce.com/john.gpg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] congested/busy on trunk?
On Wed, Mar 18, 2020 at 2:37 PM John Roman wrote: > ive enabled logging. aside from a realm error i see on my endpoint, im > still not sure whats up 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. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] congested/busy on trunk?
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
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:13107950...@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"
;tag=3166828162
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 --->
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" ;tag=3166828162
To:
;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:13107950...@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"
;tag=3166828162
To:
;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:13107950...@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"
;tag=3166828162
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:13107950...@dunkel.dev1ce.com:5060;transport=tcp",response="6c4a62c6b4061e4b9312910a974abc4b",algorithm=md5,opaque="733bad0a5366d9f2",qop=auth,cnonce="xyz",nc=0001
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@2605:e000:130a:fb:de1:71fc:e257:6f4e
From: "demo-alice" ;tag=3166828162
To:
CSeq: 8613 INVITE
Server: Asterisk PBX GIT-master-0cde95ec89
Content-Length: 0
-- Executing [13107950860@anveo_sip:1]
Dial("PJSIP/demo-alice-0002", "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-0002'
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
Re: [asterisk-users] congested/busy on trunk?
On Sat, Mar 14, 2020 at 2:02 PM John Roman 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@anveo_sip:1] Dial("PJSIP/demo-alice-0005",
> "PJSIP/blah@mytrunk") in new stack
> -- Called PJSIP/blah@mytrunk
> -- PJSIP/mytrunk-0006 is ringing
> -- PJSIP/mytrunk-0006 is ringing
> -- PJSIP/mytrunk-0006 is making progress passing it to
> PJSIP/demo-alice-0005
>> 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-0006 is making progress passing it to
> PJSIP/demo-alice-0005
> == Everyone is busy/congested at this time (1:1/0/0)
> -- Auto fallthrough, channel 'PJSIP/demo-alice-0005' 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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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-0005",
"PJSIP/blah@mytrunk") in new stack
-- Called PJSIP/blah@mytrunk
-- PJSIP/mytrunk-0006 is ringing
-- PJSIP/mytrunk-0006 is ringing
-- PJSIP/mytrunk-0006 is making progress passing it to
PJSIP/demo-alice-0005
> 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-0006 is making progress passing it to
PJSIP/demo-alice-0005
== Everyone is busy/congested at this time (1:1/0/0)
-- Auto fallthrough, channel 'PJSIP/demo-alice-0005' status is 'BUSY'
Any idea what im doing wrong? Thanks :)
--
-- -- --
j...@dev1ce.com
https://dev1ce.com/john.gpg
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Congested/busy
hi all i have a TE110P connected to my PBX when i try to call a extension number in other location 3525 the asterisk give me a error -- User entered '3525' -- Executing [EMAIL PROTECTED]:4] GotoIf(Zap/31-1, 0?6:5) in new stack -- Goto (lacnicuy,450,5) -- Executing [EMAIL PROTECTED]:5] Dial(Zap/31-1, IAX2/lacnic:[EMAIL PROTECTED]/3525|30|r) in new stack -- Called lacnic:[EMAIL PROTECTED]/3525 -- IAX2/nicbr-1 is circuit-busy [Oct 11 10:08:02] NOTICE[2763]: chan_iax2.c:2925 __auto_congest: Auto-congesting call due to slow response -- Hungup 'IAX2/nicbr-1' [Oct 11 10:08:02] NOTICE[2781]: cdr.c:434 ast_cdr_free: CDR on channel 'IAX2/nicbr-1' not posted == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:6] Hangup(Zap/31-1, ) in new stack == Spawn extension (lacnicuy, 450, 6) exited non-zero on 'Zap/31-1' -- Hungup 'Zap/31-1' anybody can help me with this? thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
