rendered paste body -- Executing [s@macro-hangupcall:9] Hangup("SIP/108-b6a44988", "") in new st
ack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/108-b6a449
88' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/108-b6a
44988'
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/108-b6
a44988' in macro 'dialout-trunk'
== Spawn extension (from-internal, 036171111, 4) exited non-zero on 'SIP/108-b
6a44988'
-- ast_get_srv: SRV lookup for '_sip._UDP.proxy.ideasip.com' mapped to host
proxy.ideasip.com, port 5060
-- ast_get_srv: SRV lookup for '_sip._UDP.proxy.ideasip.com' mapped to host
proxy.ideasip.com, port 5060
voip*CLI> set verbose
No such command 'set verbose' (type 'help set verbose' for other possible comman
ds)
voip*CLI> help set verbose
No such command 'set verbose'.
voip*CLI> sip set verbose
No such command 'sip set verbose' (type 'help sip set verbose' for other possibl
e commands)
voip*CLI> quit
[voip.mehadrin.net modules]#
[voip.mehadrin.net modules]# asterisk -r
Asterisk 1.6.0.10-FONCORE-r40, Copyright (C) 1999 - 2008 Digium, Inc. and others
.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail
s.
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 1.6.0.10-FONCORE-r40 currently running on voip (pid = 3880
)
Verbosity is at least 3
voip*CLI> sip set debug on
SIP Debugging enabled
Reliably Transmitting (NAT) to 80.74.123.8:46329:
OPTIONS sip:106@80.74.123.8:46329;rinstance=808f9d34d9c2a103 SIP/2.0
Via: SIP/2.0/UDP 91.143.227.19:5060;branch=z9hG4bK2eb96455;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@91.143.227.19>;tag=as3151e8e4
To: <sip:106@80.74.123.8:46329;rinstance=808f9d34d9c2a103>
Contact: <sip:Unknown@91.143.227.19>
Call-ID: 29be334d42747eda16bef9556be5c827@91.143.227.19
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Wed, 18 Jan 2012 15:49:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
---
voip*CLI>
<--- SIP read from UDP://80.74.123.8:46329 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.143.227.19:5060;branch=z9hG4bK2eb96455;rport=5060
Contact: <sip:10.0.0.15:2976>
To: <sip:106@80.74.123.8:46329;rinstance=808f9d34d9c2a103>;tag=b7eac84b
From: "Unknown"<sip:Unknown@91.143.227.19>;tag=as3151e8e4
Call-ID: 29be334d42747eda16bef9556be5c827@91.143.227.19
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF
O
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '29be334d42747eda16bef9556be5c827@91.143.227.19' Me
thod: OPTIONS
voip*CLI>
<--- SIP read from UDP://82.80.200.62:5060 --->
INVITE sip:0785552730@91.143.227.19 SIP/2.0
CSeq: 1 INVITE
Via: SIP/2.0/UDP 82.80.200.62:5060;branch=z9hG4bK180149121732182740893109
From: <sip:026411180@82.80.200.62:5060>;tag=1801491217322740893109
Call-ID: 89289310907e99b9dd4fdfb11d0890f1d944cc1916@82.80.200.62
To: <sip:0785552730@91.143.227.19>
Contact: <sip:82.80.200.62:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 246
v=0
o=VoipSwitch 6210 7210 IN IP4 82.80.200.62
s=VoipSIP
i=Audio Session
c=IN IP4 82.80.200.62
t=0 0
m=audio 6210 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (9 headers 12 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
Sending to 82.80.200.62 : 5060 (no NAT)
Using INVITE request as basis request - 89289310907e99b9dd4fdfb11d0890f1d944cc19
16@82.80.200.62
No user '026411180' in SIP users list
Found peer 'SipMe' for '026411180' from 82.80.200.62:5060
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 82.80.200.62:6210
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x28010c (ulaw|alaw|g729|h263|h264), peer - audio=0x100 (g729
)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone
-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 82.80.200.62:6210
Peer video RTP is at port 82.80.200.62:12600
Looking for 0785552730 in from-sip-external (domain 91.143.227.19)
list_route: hop: <sip:82.80.200.62:5060;transport=udp>
<--- Transmitting (no NAT) to 82.80.200.62:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.80.200.62:5060;branch=z9hG4bK180149121732182740893109;receiv
ed=82.80.200.62
From: <sip:026411180@82.80.200.62:5060>;tag=1801491217322740893109
To: <sip:0785552730@91.143.227.19>
Call-ID: 89289310907e99b9dd4fdfb11d0890f1d944cc1916@82.80.200.62
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:0785552730@91.143.227.19>
Content-Length: 0
<------------>
-- Executing [0785552730@from-sip-external:1] NoOp("SIP/0785552730-b6a274a0"
, "Received incoming SIP connection from unknown peer to 0785552730") in new sta
ck
-- Executing [0785552730@from-sip-external:2] Set("SIP/0785552730-b6a274a0",
"DID=0785552730") in new stack
-- Executing [0785552730@from-sip-external:3] Goto("SIP/0785552730-b6a274a0"
, "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/0785552730-b6a274a0", "0?fr
om-trunk,0785552730,1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/0785552730-b6a274a0", "TIMEOUT
(absolute)=15") in new stack
Channel will hangup at 2012-01-18 12:50:03.000 GMT+3.
-- Executing [s@from-sip-external:3] Answer("SIP/0785552730-b6a274a0", "") i
n new stack
Audio is at 91.143.227.19 port 17234
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
voip*CLI>
<--- Reliably Transmitting (no NAT) to 82.80.200.62:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.80.200.62:5060;branch=z9hG4bK180149121732182740893109;receiv
ed=82.80.200.62
From: <sip:026411180@82.80.200.62:5060>;tag=1801491217322740893109
To: <sip:0785552730@91.143.227.19>;tag=as12ae55de
Call-ID: 89289310907e99b9dd4fdfb11d0890f1d944cc1916@82.80.200.62
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:0785552730@91.143.227.19>
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 126067678 126067678 IN IP4 91.143.227.19
s=Asterisk PBX 1.6.0.10-FONCORE-r40
c=IN IP4 91.143.227.19
t=0 0
m=audio 17234 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
voip*CLI>
<--- SIP read from UDP://82.80.200.62:5060 --->
ACK sip:0785552730@91.143.227.19 SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP 82.80.200.62:5060;branch=z9hG4bK180149121732182740893125
From: <sip:026411180@82.80.200.62:5060>;tag=1801491217322740893109
Call-ID: 89289310907e99b9dd4fdfb11d0890f1d944cc1916@82.80.200.62
To: <sip:0785552730@91.143.227.19>;tag=as12ae55de
Contact: <sip:82.80.200.62:5060;transport=udp>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
-- Executing [s@from-sip-external:4] Wait("SIP/0785552730-b6a274a0", "2") in
new stack
-- Executing [s@from-sip-external:5] Playback("SIP/0785552730-b6a274a0", "ss
-noservice") in new stack
-- <SIP/0785552730-b6a274a0> Playing 'ss-noservice.gsm' (language 'en')
voip*CLI>
<--- SIP read from UDP://80.74.123.8:46329 --->
<------------->
-- Executing [s@from-sip-external:6] PlayTones("SIP/0785552730-b6a274a0", "c
ongestion") in new stack
-- Executing [s@from-sip-external:7] Congestion("SIP/0785552730-b6a274a0", "
5") in new stack
voip*CLI>
<--- SIP read from UDP://82.80.200.62:5060 --->
BYE sip:0785552730@91.143.227.19 SIP/2.0
CSeq: 2 BYE
Via: SIP/2.0/UDP 82.80.200.62:5060;branch=z9hG4bK180149121741182740901390
From: <sip:026411180@82.80.200.62:5060>;tag=1801491217322740893109
Call-ID: 89289310907e99b9dd4fdfb11d0890f1d944cc1916@82.80.200.62
To: <sip:0785552730@91.143.227.19>;tag=as12ae55de
Contact: <sip:82.80.200.62:5060;transport=udp>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Sending to 82.80.200.62 : 5060 (no NAT)
<--- Transmitting (no NAT) to 82.80.200.62:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.80.200.62:5060;branch=z9hG4bK180149121741182740901390;receiv
ed=82.80.200.62
From: <sip:026411180@82.80.200.62:5060>;tag=1801491217322740893109
To: <sip:0785552730@91.143.227.19>;tag=as12ae55de
Call-ID: 89289310907e99b9dd4fdfb11d0890f1d944cc1916@82.80.200.62
CSeq: 2 BYE
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/078555273
0-b6a274a0'
-- Executing [h@from-sip-external:1] NoOp("SIP/0785552730-b6a274a0", "Hangup
") in new stack
-- Executing [h@from-sip-external:2] Set("SIP/0785552730-b6a274a0", "DID=s")
in new stack
-- Executing [h@from-sip-external:3] Goto("SIP/0785552730-b6a274a0", "s,1")
in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/0785552730-b6a274a0", "0?fr
om-trunk,s,1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/0785552730-b6a274a0", "TIMEOUT
(absolute)=15") in new stack
Channel will hangup at 2012-01-18 12:50:12.000 GMT+3.
-- Executing [s@from-sip-external:3] Answer("SIP/0785552730-b6a274a0", "") i
n new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/078555273
0-b6a274a0'
Really destroying SIP dialog '89289310907e99b9dd4fdfb11d0890f1d944cc1916@82.80.2
00.62' Method: BYE
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 91.143.227.19:5060;branch=z9hG4bK018a3e65;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@91.143.227.19>;tag=as67d024e5
To: <sip:ekiga.net>
Contact: <sip:Unknown@91.143.227.19>
Call-ID: 4ac9bc410330b86a3e2f089e60b50a72@91.143.227.19
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Wed, 18 Jan 2012 15:49:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
---
Reliably Transmitting (no NAT) to 86.64.162.35:5060:
OPTIONS sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 91.143.227.19:5060;branch=z9hG4bK4c441bb2;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@91.143.227.19>;tag=as73dff7c9
To: <sip:ekiga.net>
Contact: <sip:Unknown@91.143.227.19>
Call-ID: 57e5e206270cb77056365ebb0ba1d467@91.143.227.19
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Wed, 18 Jan 2012 15:49:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
---
voip*CLI>
<--- SIP read from UDP://86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.143.227.19:5060;branch=z9hG4bK018a3e65;rport=5060
From: "Unknown" <sip:Unknown@91.143.227.19>;tag=as67d024e5
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.f942
Call-ID: 4ac9bc410330b86a3e2f089e60b50a72@91.143.227.19
CSeq: 102 OPTIONS
Server: Kamailio (1.5.3-notls (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4ac9bc410330b86a3e2f089e60b50a72@91.143.227.19' Me
thod: OPTIONS
voip*CLI>
<--- SIP read from UDP://86.64.162.35:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.143.227.19:5060;branch=z9hG4bK4c441bb2;rport=5060
From: "Unknown" <sip:Unknown@91.143.227.19>;tag=as73dff7c9
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.ca06
Call-ID: 57e5e206270cb77056365ebb0ba1d467@91.143.227.19
CSeq: 102 OPTIONS
Server: Kamailio (1.5.3-notls (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '57e5e206270cb77056365ebb0ba1d467@91.143.227.19' Me
thod: OPTIONS
voip*CLI> sip set debug off
SIP Debugging Disabled
voip*CLI>