Channel: SIP/sipnet/###########
krll: ~ $ sudo asterisk -rvvv
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Asterisk 11.7.0~dfsg-1ubuntu1, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@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 11.7.0~dfsg-1ubuntu1 currently running on krll (pid = 1076)
krll*CLI> sip set debug peer sipnet
SIP Debugging Enabled for IP: 212.53.40.40
krll*CLI> channel originate SIP/sipnet/89991628348 application hangup
== Using SIP RTP CoS mark 5
Audio is at 10358
Adding codec 100008 (g729) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
INVITE sip:89991628348@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.235.107:5060;branch=z9hG4bK5b00a0c9
Max-Forwards: 70
From: "Anonymous" <sip:0042138651@anonymous.invalid>;tag=as60061e20
To: <sip:89991628348@sipnet.ru>
Contact: <sip:0042138651@10.0.235.107:5060>
Call-ID: 0365e01c79ab04dd1cc1c9e90a70250c@sipnet.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Date: Wed, 24 Feb 2016 15:14:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 385
v=0
o=root 596327080 596327080 IN IP4 10.0.235.107
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 10.0.235.107
t=0 0
m=audio 10358 RTP/AVP 18 4 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.235.107:5060;branch=z9hG4bK5b00a0c9;received=52.50.51.240
From: "Anonymous" <sip:0042138651@anonymous.invalid>;tag=as60061e20
To: <sip:89991628348@sipnet.ru>
Call-ID: 0365e01c79ab04dd1cc1c9e90a70250c@sipnet.ru
CSeq: 102 INVITE
Server: CommuniGatePro/6.1.8
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 401 Authentication required
Via: SIP/2.0/UDP 10.0.235.107:5060;branch=z9hG4bK5b00a0c9;received=52.50.51.240
From: "Anonymous" <sip:0042138651@anonymous.invalid>;tag=as60061e20
To: <sip:89991628348@sipnet.ru>;tag=D59F540D
Call-ID: 0365e01c79ab04dd1cc1c9e90a70250c@sipnet.ru
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="etc.tario.ru",nonce="4F3E806CA40CE71F9697",opaque="opaq",qop="auth",algorithm=MD5
Server: CommuniGatePro/6.1.8
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:89991628348@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.235.107:5060;branch=z9hG4bK5b00a0c9
Max-Forwards: 70
From: "Anonymous" <sip:0042138651@anonymous.invalid>;tag=as60061e20
To: <sip:89991628348@sipnet.ru>;tag=D59F540D
Contact: <sip:0042138651@10.0.235.107:5060>
Call-ID: 0365e01c79ab04dd1cc1c9e90a70250c@sipnet.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0
---
Audio is at 10358
Adding codec 100008 (g729) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
INVITE sip:89991628348@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.235.107:5060;branch=z9hG4bK1dc9faa3
Max-Forwards: 70
From: "Anonymous" <sip:0042138651@anonymous.invalid>;tag=as60061e20
To: <sip:89991628348@sipnet.ru>
Contact: <sip:0042138651@10.0.235.107:5060>
Call-ID: 0365e01c79ab04dd1cc1c9e90a70250c@sipnet.ru
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Authorization: Digest username="webtv", realm="etc.tario.ru", algorithm=MD5, uri="sip:89991628348@sipnet.ru", nonce="4F3E806CA40CE71F9697", response="432af14a4b99f735bc818624e6ec5b8d", opaque="opaq
", qop=auth, cnonce="3fc42f62", nc=00000001
Date: Wed, 24 Feb 2016 15:14:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 385
v=0
o=root 596327080 596327081 IN IP4 10.0.235.107
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 10.0.235.107
t=0 0
m=audio 10358 RTP/AVP 18 4 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 500 account has been moved to a remote system
Via: SIP/2.0/UDP 10.0.235.107:5060;branch=z9hG4bK1dc9faa3;received=52.50.51.240
From: "Anonymous" <sip:0042138651@anonymous.invalid>;tag=as60061e20
To: <sip:89991628348@sipnet.ru>;tag=D739A71D
Call-ID: 0365e01c79ab04dd1cc1c9e90a70250c@sipnet.ru
CSeq: 103 INVITE
Server: CommuniGatePro/6.1.8
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
-- Got SIP response 500 "account has been moved to a remote system" back from 212.53.40.40:5060
Transmitting (no NAT) to 212.53.40.40:5060:
ACK sip:89991628348@sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.235.107:5060;branch=z9hG4bK1dc9faa3
Max-Forwards: 70
From: "Anonymous" <sip:0042138651@anonymous.invalid>;tag=as60061e20
To: <sip:89991628348@sipnet.ru>;tag=D739A71D
Contact: <sip:0042138651@10.0.235.107:5060>
Call-ID: 0365e01c79ab04dd1cc1c9e90a70250c@sipnet.ru
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Content-Length: 0
---
Really destroying SIP dialog '0365e01c79ab04dd1cc1c9e90a70250c@sipnet.ru' Method: INVITE
[Feb 24 15:14:59] NOTICE[1312]: chan_sip.c:15023 sip_reregister: -- Re-registration for 0042138651@sipnet.ru
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 212.53.40.40:5060:
REGISTER sip:sipnet.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.235.107:5060;branch=z9hG4bK6fe31b23
Max-Forwards: 70
From: <sip:0042138651@sipnet.ru>;tag=as209a742b
To: <sip:0042138651@sipnet.ru>
Call-ID: 275dac38673d3aef18826e3c63d34d25@127.0.0.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1
Authorization: Digest username="0042138651", realm="etc.tario.ru", algorithm=MD5, uri="sip:sipnet.ru", nonce="2A0A90163B878722085A", response="bdf13ab8edda50ea71c2bdde7f4268ad", opaque="opaq", qop=
auth, cnonce="6851dd62", nc=00000002
Expires: 120
Contact: <sip:0042138651@10.0.235.107:5060>
Content-Length: 0
---
<--- SIP read from UDP:212.53.40.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.235.107:5060;branch=z9hG4bK6fe31b23;received=52.50.51.240
Path: <sip:nudp-t52.50.51.240-5060s212.53.40.40-5060.u.cgatepro;lr>
From: <sip:0042138651@sipnet.ru>;tag=as209a742b
To: <sip:0042138651@sipnet.ru>;tag=DC6DA014
Call-ID: 275dac38673d3aef18826e3c63d34d25@127.0.0.1
CSeq: 104 REGISTER
Expires: 116
Contact: <sip:0042138651@10.0.235.107:5060>;expires=116
Event: registration
Date: Wed, 24 Feb 2016 15:14:59 GMT
Allow: PUBLISH,SUBSCRIBE
Supported: path,gruu
Allow-Events: presence,message-summary,reg,dialog,line-seize,keep-alive,refer
Server: CommuniGatePro/6.1.8
Content-Length: 0
<------------->
--- (16 headers 0 lines) ---
Scheduling destruction of SIP dialog '275dac38673d3aef18826e3c63d34d25@127.0.0.1' in 32000 ms (Method: REGISTER)
[Feb 24 15:14:59] NOTICE[1312]: chan_sip.c:23472 handle_response_register: Outbound Registration: Expiry for sipnet.ru is 116 sec (Scheduling reregistration in 101 s)
<--- SIP read from UDP:212.53.40.40:5060 --->
<------------->
Really destroying SIP dialog '275dac38673d3aef18826e3c63d34d25@127.0.0.1' Method: REGISTER
krll*CLI>
sudo tail -n 20 /var/log/asterisk/messages
[Feb 22 14:50:56] ERROR[1711] res_calendar_ews.c: Exchange Web Service calendar module require
neon >= 0.29.1, but neon 0.30.0: Library build, IPv6, libxml 2.9.1, zlib 1.2.8, GNU TLS 2.12.
23. is installed.
[Feb 22 14:50:56] NOTICE[1711] cel_custom.c: No mappings found in cel_custom.conf. Not logging
CEL to custom CSVs.
[Feb 22 14:50:56] WARNING[1711] cel_pgsql.c: CEL pgsql config file missing global section.
[Feb 22 14:50:56] NOTICE[1711] cdr_pgsql.c: cdr_pgsql configuration contains no global section
, skipping module load.
[Feb 22 14:50:56] NOTICE[1711] cel_tds.c: cel_tds has no global category, nothing to configure
.
[Feb 22 14:50:56] WARNING[1711] cel_tds.c: cel_tds module had config problems; declining load
[Feb 22 14:50:56] NOTICE[1711] pbx_ael.c: Starting AEL load process.
[Feb 22 14:50:56] NOTICE[1711] pbx_ael.c: AEL load process: parsed config file name '/etc/aste
risk/extensions.ael'.
[Feb 22 14:50:56] NOTICE[1711] pbx_ael.c: AEL load process: checked config file name '/etc/ast
erisk/extensions.ael'.
[Feb 22 14:50:56] NOTICE[1711] pbx_ael.c: AEL load process: compiled config file name '/etc/as
terisk/extensions.ael'.
[Feb 22 14:50:56] NOTICE[1711] pbx_ael.c: AEL load process: merged config file name '/etc/aste
risk/extensions.ael'.
[Feb 22 14:50:56] NOTICE[1711] pbx_ael.c: AEL load process: verified config file name '/etc/as
terisk/extensions.ael'.
[Feb 22 14:50:56] WARNING[1711] pbx_dundi.c: Unable to look up host 'krll'
[Feb 22 14:51:04] NOTICE[1728][C-00000000] chan_sip.c: Failed to authenticate on INVITE to '<s
ip:0042138651@sipnet.ru>;tag=as051657a8'
[Feb 22 14:51:04] NOTICE[1753] pbx_spool.c: Call failed to go through, reason (8) Congestion (
circuits busy)
[Feb 22 14:51:24] NOTICE[1728][C-00000001] chan_sip.c: Failed to authenticate on INVITE to '<s
ip:0042138651@sipnet.ru>;tag=as0ccca134'
[Feb 22 14:51:24] NOTICE[1759] pbx_spool.c: Call failed to go through, reason (8) Congestion (
circuits busy)
[Feb 22 14:51:30] NOTICE[1728][C-00000002] chan_sip.c: Failed to authenticate on INVITE to '<s
ip:0042138651@sipnet.ru>;tag=as36b20ddd'
[Feb 22 14:51:30] NOTICE[1760] pbx_spool.c: Call failed to go through, reason (8) Congestion (
circuits busy)
[Feb 22 14:51:30] NOTICE[1760] pbx_spool.c: Queued call to SIP/sipnet/89600636463 expired with
out completion after 2 attempts