[asterisk-bugs] [Asterisk 0018016]: SIP Trunk ${DIALSTATUS} wrong return code - it is always "ANSWER" status
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Sep 21 04:48:18 CDT 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18016
======================================================================
Reported By: romirikos
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 18016
Category: Channels/chan_sip/NewFeature
Reproducibility: have not tried
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.8.0-beta5
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-09-20 03:30 CDT
Last Modified: 2010-09-21 04:48 CDT
======================================================================
Summary: SIP Trunk ${DIALSTATUS} wrong return code - it is
always "ANSWER" status
Description:
1. First I show my SIP Trunk and SIP phone user configuration
1.1 SIP Trunk configuration
[WMAG006-trunk]
dtmfmode=rfc2833
type=friend
host=*** IP VoIP-Provider ***
fromuser=*** My username ***
fromdomain=*** VoIP-Provider Domain Name ***
username=*** username ***
secret=*** password ***
insecure=port,invite
canreinvite=no
qualify=yes
callgroup=
context=WMAG006-outbounds-calls
nat=yes
disallow=all
allow=g729
1.2 SIP phone username configuration
[505]
allow=all
secret=505
dtmfmode=rfc2833
canreinvite=no
context=WMAG006-outbounds-calls
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
dial=SIP/505
permit=0.0.0.0/0.0.0.0
callerid=505
call-limit=50
faxdetect=no
disallow=all
allow=g729
2. Here is my DialPlan in /etc/asterisk/extensions.conf
[WMAG006-outbounds-calls]
; England, Spain (Code number + phone number = 12 digits)
; exten => _XXXXXXXXXXXX,1,Set(CALLERID(all)=08000963317)
; exten => _XXXXXXXXXXXX,1,Dial(SIP/${EXTEN}@WMAG006-trunk,25,g)
exten => _XXXXXXXXXXXX,1,Dial(SIP/WMAG006-trunk/${EXTEN},5,R)
exten => _XXXXXXXXXXXX,n,Hangup()
; England 13 digits
;exten => _XXXXXXXXXXXXX,1,Dial(SIP/${EXTEN}@WMAG006-trunk,5)
;exten => _XXXXXXXXXXXXX,n,Hangup()
; USA, Australia (Code number + phone number = 11 digits)
;exten => _XXXXXXXXXXX,1,Dial(SIP/${EXTEN}@WMAG006-trunk,5)
;exten => _XXXXXXXXXXX,n,Hangup()
; Denmark (Code number + phone number = 10 digits)
;exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@WMAG006-trunk,5)
;exten => _XXXXXXXXXX,n,Hangup()
exten => 505,1,Answer()
exten => 505,n,Dial(SIP/505,25)
exten => 505,n,Hangup()
exten => 501,1,Answer()
exten => 501,n,Dial(SIP/501,25)
exten => 501,n,Hangup()
exten => h,1,Goto(s-{DIALSTATUS},1)
; exten => h,1,System(/bin/sh -c "/bin/echo ${DIALSTATUS} >
/usr/local/asterisk/wmag006/status")
exten => s-CANCEL,1,System(/bin/sh -c "/bin/echo Cancel >
/usr/local/asterisk/wmag006/status")
exten => s-ANSWER,1,System(/bin/sh -c "/bin/echo Answer >
/usr/local/asterisk/wmag006/status")
exten => s-NOANSWER,1,System(/bin/sh -c "/bin/echo NoAnswer >
/usr/local/asterisk/wmag006/status")
exten => s-BUSY,1,System(/bin/sh -c "/bin/echo Busy `date +%F-%H:%M` >
/usr/local/asterisk/wmag006/status")
exten => s-CONGESTION,1,System(/bin/sh -c "/bin/echo Congestion `date
+%F-%H:%M` > /usr/local/asterisk/wmag006/status")
exten => s-CHANUNAVAIL,1,System(/bin/sh -c "/bin/echo Chanunavail `date
+%F-%H:%M` > /usr/local/asterisk/wmag006/status")
3. Debug output from Asterisk 1.8 CLI
-- Registered SIP '505' at 192.168.0.7:2950
> Saved useragent "sipLite" for peer 505
[Sep 20 11:30:58] NOTICE[30357]: chan_sip.c:19477
handle_response_peerpoke: Peer '505' is now Reachable. (24ms / 2000ms)
== Using SIP RTP CoS mark 5
-- Executing [380623880525 at WMAG006-outbounds-calls:1]
Dial("SIP/505-00000002", "SIP/WMAG006-trunk/380623880525,5,R") in new
stack
== Using SIP RTP CoS mark 5
-- Called WMAG006-trunk/380623880525
-- SIP/WMAG006-trunk-00000003 answered SIP/505-00000002
-- Locally bridging SIP/505-00000002 and SIP/WMAG006-trunk-00000003
-- Executing [h at WMAG006-outbounds-calls:1] System("SIP/505-00000002",
"/bin/sh -c "/bin/echo ANSWER > /usr/local/asterisk/wmag006/status"") in
new stack
== Spawn extension (WMAG006-outbounds-calls, 380623880525, 1) exited
non-zero on 'SIP/505-00000002'
-------------------------------------------------------------------------
So, the issue is when I place the call through the SIP trunk it's always
return - "ANSWER" status even if the other side was BUSY, NOANSWER,
CONGESTION and etc.
But if I place the call between (Local IP phones that connected directly
to Asterisk 1.8) for example I call from 505 to 501 and it return RIGHT
${DIALSTATUS} code
======================================================================
----------------------------------------------------------------------
(0127181) romirikos (reporter) - 2010-09-21 04:48
https://issues.asterisk.org/view.php?id=18016#c127181
----------------------------------------------------------------------
This is Full SIP Debug output
SIP Debugging enabled
Reliably Transmitting (NAT) to 192.168.2.27:5060:
OPTIONS sip:503 at 192.168.2.27:5060;line=3aehmciw SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK6b21a27b;rport
From: "asterisk" <sip:asterisk at 192.168.2.1>;tag=as1ec1276f
To: <sip:503 at 192.168.2.27:5060;line=3aehmciw>
Contact: <sip:asterisk at 192.168.2.1>
Call-ID: 589b67ab6b35dd0f33901fd56ea54d54 at 192.168.2.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 21 Sep 2010 09:39:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
---
<--- SIP read from 192.168.2.27:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK6b21a27b;rport=5060
f: "asterisk" <sip:asterisk at 192.168.2.1>;tag=as1ec1276f
t: <sip:503 at 192.168.2.27:5060;line=3aehmciw>
i: 589b67ab6b35dd0f33901fd56ea54d54 at 192.168.2.1
CSeq: 102 OPTIONS
m: <sip:27 at 192.168.2.27:5060;line=ydg4eu9f>;flow-id=1
User-Agent: snom320/6.5.4
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
l: 0
<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog
'589b67ab6b35dd0f33901fd56ea54d54 at 192.168.2.1' Method: OPTIONS
<--- SIP read from 192.168.2.23:5060 --->
INVITE sip:380623880525 at 195.184.196.222 SIP/2.0
v: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-9fqz7t2ezr3j;rport
f: "505" <sip:505 at 195.184.196.222>;tag=0oc4ee71xr
t: <sip:380623880525 at 195.184.196.222>
i: 3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
CSeq: 1 INVITE
Max-Forwards: 30
m: <sip:505 at 192.168.2.23:5060;line=x0v8xene>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/6.2.2
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
c: application/sdp
l: 475
v=0
o=root 1120826698 1120826698 IN IP4 192.168.2.23
s=call
c=IN IP4 192.168.2.23
t=0 0
m=audio 22426 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:btyoi9sVrX6ST6gnWRVRMmff/STy+Xbvc+in/h+s
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv
<------------->
--- (18 headers 19 lines) ---
Sending to 192.168.2.23 : 5060 (no NAT)
Using INVITE request as basis request -
3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
<--- Reliably Transmitting (NAT) to 192.168.2.23:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.2.23:5060;branch=z9hG4bK-9fqz7t2ezr3j;received=192.168.2.23;rport=5060
From: "505" <sip:505 at 195.184.196.222>;tag=0oc4ee71xr
To: <sip:380623880525 at 195.184.196.222>;tag=as69eb4b51
Call-ID: 3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="1d4ab296"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'3c267b928b29-b7et3f97hqmi at snom320-0004132485B0' in 32000 ms (Method:
INVITE)
Found user '505'
<--- SIP read from 192.168.2.23:5060 --->
ACK sip:380623880525 at 195.184.196.222 SIP/2.0
v: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-9fqz7t2ezr3j;rport
f: "505" <sip:505 at 195.184.196.222>;tag=0oc4ee71xr
t: <sip:380623880525 at 195.184.196.222>;tag=as69eb4b51
i: 3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
CSeq: 1 ACK
Max-Forwards: 30
m: <sip:505 at 192.168.2.23:5060;line=x0v8xene>;flow-id=1
l: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from 192.168.2.23:5060 --->
INVITE sip:380623880525 at 195.184.196.222 SIP/2.0
v: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-9ov4p7niuvbq;rport
f: "505" <sip:505 at 195.184.196.222>;tag=0oc4ee71xr
t: <sip:380623880525 at 195.184.196.222>
i: 3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
CSeq: 2 INVITE
Max-Forwards: 30
m: <sip:505 at 192.168.2.23:5060;line=x0v8xene>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/6.2.2
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest
username="505",realm="asterisk",nonce="1d4ab296",uri="sip:380623880525 at 195.184.196.222",response="28ec0950e18d05f656a39e57d30cdd11",algorithm=md5
c: application/sdp
l: 475
v=0
o=root 1120826698 1120826698 IN IP4 192.168.2.23
s=call
c=IN IP4 192.168.2.23
t=0 0
m=audio 22426 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:btyoi9sVrX6ST6gnWRVRMmff/STy+Xbvc+in/h+s
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv
<------------->
--- (19 headers 19 lines) ---
Sending to 192.168.2.23 : 5060 (NAT)
Using INVITE request as basis request -
3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
Found user '505'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Found audio description format pcmu for ID 0
Found audio description format pcma for ID 8
Found audio description format g722 for ID 9
Found audio description format g726-32 for ID 2
Found audio description format gsm for ID 3
Found audio description format g729 for ID 18
Found audio description format g723 for ID 4
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x190f
(g723|gsm|ulaw|alaw|g726|g729|g722)/video=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 192.168.2.23:22426
Looking for 380623880525 in WMAG006-outbounds-calls (domain
195.184.196.222)
list_route: hop: <sip:505 at 192.168.2.23:5060;line=x0v8xene>
<--- Transmitting (NAT) to 192.168.2.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.23:5060;branch=z9hG4bK-9ov4p7niuvbq;received=192.168.2.23;rport=5060
From: "505" <sip:505 at 195.184.196.222>;tag=0oc4ee71xr
To: <sip:380623880525 at 195.184.196.222>
Call-ID: 3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:380623880525 at 192.168.2.1>
Content-Length: 0
<------------>
-- Executing [380623880525 at WMAG006-outbounds-calls:1]
Dial("SIP/505-00000008", "SIP/380623880525 at WMAG006-trunk") in new stack
Audio is at 195.184.196.222 port 16174
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 213.144.184.12:5060:
INVITE sip:380623880525 at foib.future-b.eu SIP/2.0
Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK00870a6a;rport
From: "505" <sip:WMAG006 at foib.future-b.eu>;tag=as51bae2b2
To: <sip:380623880525 at foib.future-b.eu>
Contact: <sip:WMAG006 at 195.184.196.222>
Call-ID: 356e350427ca0d40294efae962c045f5 at foib.future-b.eu
CSeq: 102 INVITE
ser-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 21 Sep 2010 09:40:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 17187 17187 IN IP4 195.184.196.222
s=session
c=IN IP4 195.184.196.222
t=0 0
m=audio 16174 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
---
-- Called 380623880525 at WMAG006-trunk
<--- SIP read from 213.144.184.12:5060 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK00870a6a;rport
From: "505" <sip:WMAG006 at foib.future-b.eu>;tag=as51bae2b2
Call-ID: 356e350427ca0d40294efae962c045f5 at foib.future-b.eu
To: <sip:380623880525 at foib.future-b.eu>
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from 213.144.184.12:5060 --->
SIP/2.0 407 Proxy Authentication Required
CSeq: 102 INVITE
Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK00870a6a;rport
From: "505" <sip:WMAG006 at foib.future-b.eu>;tag=as51bae2b2
Call-ID: 356e350427ca0d40294efae962c045f5 at foib.future-b.eu
To: <sip:380623880525 at foib.future-b.eu>;tag=1037221576942191
Proxy-Authenticate: DIGEST realm="VoipSwitch",
nonce="15769421917680812821101026409372238238"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 213.144.184.12:5060:
ACK sip:380623880525 at foib.future-b.eu SIP/2.0
Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK00870a6a;rport
From: "505" <sip:WMAG006 at foib.future-b.eu>;tag=as51bae2b2
To: <sip:380623880525 at foib.future-b.eu>;tag=1037221576942191
Contact: <sip:WMAG006 at 195.184.196.222>
Call-ID: 356e350427ca0d40294efae962c045f5 at foib.future-b.eu
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Audio is at 195.184.196.222 port 16174
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 213.144.184.12:5060:
INVITE sip:380623880525 at foib.future-b.eu SIP/2.0
Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK07fef3d8;rport
From: "505" <sip:WMAG006 at foib.future-b.eu>;tag=as51bae2b2
To: <sip:380623880525 at foib.future-b.eu>
Contact: <sip:WMAG006 at 195.184.196.222>
Call-ID: 356e350427ca0d40294efae962c045f5 at foib.future-b.eu
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="WMAG006", realm="VoipSwitch",
algorithm=MD5, uri="sip:380623880525 at foib.future-b.eu",
nonce="15769421917680812821101026409372238238",
response="bd0278f6c9f6cb162d92abefa36ac70f"
Date: Tue, 21 Sep 2010 09:40:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 17187 17188 IN IP4 195.184.196.222
s=session
c=IN IP4 195.184.196.222
t=0 0
m=audio 16174 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
---
<--- SIP read from 213.144.184.12:5060 --->
SIP/2.0 100 Trying
CSeq: 103 INVITE
Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK07fef3d8;rport
From: "505" <sip:WMAG006 at foib.future-b.eu>;tag=as51bae2b2
Call-ID: 356e350427ca0d40294efae962c045f5 at foib.future-b.eu
To: <sip:380623880525 at foib.future-b.eu>
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from 213.144.184.12:5060 --->
SIP/2.0 200 OK
CSeq: 103 INVITE
Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK07fef3d8;rport
From: "505" <sip:WMAG006 at foib.future-b.eu>;tag=as51bae2b2
Call-ID: 356e350427ca0d40294efae962c045f5 at foib.future-b.eu
To: <sip:380623880525 at foib.future-b.eu>;tag=1037221576942347
Contact: <sip:213.144.184.12:5060;transport=udp>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, NOTIFY, MESSAGE,
SUBSCRIBE, REFER
Content-Type: application/sdp
Content-Length: 236
v=0
o=- 1654832215 1576942191 IN IP4 213.144.184.12
s=VoipSIP
c=IN IP4 213.144.184.12
t=0 0
m=audio 6032 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
<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=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 213.144.184.12:6032
list_route: hop: <sip:213.144.184.12:5060;transport=udp>
set_destination: Parsing <sip:213.144.184.12:5060;transport=udp> for
address/port to send to
set_destination: set destination to 213.144.184.12, port 5060
Transmitting (NAT) to 213.144.184.12:5060:
ACK sip:213.144.184.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK43d609a2;rport
From: "505" <sip:WMAG006 at foib.future-b.eu>;tag=as51bae2b2
To: <sip:380623880525 at foib.future-b.eu>;tag=1037221576942347
Contact: <sip:WMAG006 at 195.184.196.222>
Call-ID: 356e350427ca0d40294efae962c045f5 at foib.future-b.eu
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/WMAG006-trunk-00000009 answered SIP/505-00000008
Audio is at 192.168.2.1 port 17028
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 192.168.2.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.23:5060;branch=z9hG4bK-9ov4p7niuvbq;received=192.168.2.23;rport=5060
From: "505" <sip:505 at 195.184.196.222>;tag=0oc4ee71xr
To: <sip:380623880525 at 195.184.196.222>;tag=as0db2ebea
Call-ID: 3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:380623880525 at 192.168.2.1>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 17187 17187 IN IP4 192.168.2.1
s=session
c=IN IP4 192.168.2.1
t=0 0
m=audio 17028 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
<------------>
-- Packet2Packet bridging SIP/505-00000008 and
SIP/WMAG006-trunk-00000009
<--- SIP read from 192.168.2.23:5060 --->
ACK sip:380623880525 at 192.168.2.1 SIP/2.0
v: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-10bi0abz0u6e;rport
f: "505" <sip:505 at 195.184.196.222>;tag=0oc4ee71xr
t: <sip:380623880525 at 195.184.196.222>;tag=as0db2ebea
i: 3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
CSeq: 2 ACK
Max-Forwards: 30
m: <sip:505 at 192.168.2.23:5060;line=x0v8xene>;flow-id=1
l: 0
<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (NAT) to 192.168.2.11:5060:
OPTIONS sip:504 at 192.168.2.11:5060;line=ty8vwu7u SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK31b18e1c;rport
From: "asterisk" <sip:asterisk at 192.168.2.1>;tag=as42b34502
To: <sip:504 at 192.168.2.11:5060;line=ty8vwu7u>
Contact: <sip:asterisk at 192.168.2.1>
Call-ID: 36e3b8fe6974a432044047a76a834d6b at 192.168.2.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 21 Sep 2010 09:40:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
---
<--- SIP read from 192.168.2.11:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK31b18e1c;rport=5060
f: "asterisk" <sip:asterisk at 192.168.2.1>;tag=as42b34502
t: <sip:504 at 192.168.2.11:5060;line=ty8vwu7u>
i: 36e3b8fe6974a432044047a76a834d6b at 192.168.2.1
CSeq: 102 OPTIONS
m: <sip:11 at 192.168.2.11:5060;line=lhdez2b1>;flow-id=1
User-Agent: snom320/6.2.2
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
l: 0
<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog
'36e3b8fe6974a432044047a76a834d6b at 192.168.2.1' Method: OPTIONS
<--- SIP read from 192.168.2.23:5060 --->
BYE sip:380623880525 at 192.168.2.1 SIP/2.0
v: SIP/2.0/UDP 192.168.2.23:5060;branch=z9hG4bK-wld78hghc563;rport
f: "505" <sip:505 at 195.184.196.222>;tag=0oc4ee71xr
t: <sip:380623880525 at 195.184.196.222>;tag=as0db2ebea
i: 3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
CSeq: 3 BYE
Max-Forwards: 30
m: <sip:505 at 192.168.2.23:5060;line=x0v8xene>;flow-id=1
User-Agent: snom320/6.2.2
RTP-RxStat:
Total_Rx_Pkts=592,Rx_Pkts=592,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=722,Tx_Pkts=722,Remote_Tx_Pkts=0
l: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.2.23 : 5060 (NAT)
<--- Transmitting (NAT) to 192.168.2.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.23:5060;branch=z9hG4bK-wld78hghc563;received=192.168.2.23;rport=5060
From: "505" <sip:505 at 195.184.196.222>;tag=0oc4ee71xr
To: <sip:380623880525 at 195.184.196.222>;tag=as0db2ebea
Call-ID: 3c267b928b29-b7et3f97hqmi at snom320-0004132485B0
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------>
-- Executing [h at WMAG006-outbounds-calls:1] System("SIP/505-00000008",
"/bin/sh -c "/bin/echo ANSWER > /usr/local/asterisk/wmag006/status"") in
new stack
Scheduling destruction of SIP dialog
'356e350427ca0d40294efae962c045f5 at foib.future-b.eu' in 9152 ms (Method:
INVITE)
set_destination: Parsing <sip:213.144.184.12:5060;transport=udp> for
address/port to send to
set_destination: set destination to 213.144.184.12, port 5060
Reliably Transmitting (NAT) to 213.144.184.12:5060:
BYE sip:213.144.184.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK325599a1;rport
From: "505" <sip:WMAG006 at foib.future-b.eu>;tag=as51bae2b2
To: <sip:380623880525 at foib.future-b.eu>;tag=1037221576942347
Call-ID: 356e350427ca0d40294efae962c045f5 at foib.future-b.eu
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="WMAG006", realm="VoipSwitch",
algorithm=MD5, uri="sip:213.144.184.12:5060",
nonce="15769421917680812821101026409372238238",
response="adf62a8a2498997b44166566a3e70f5b"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
== Spawn extension (WMAG006-outbounds-calls, 380623880525, 1) exited
non-zero on 'SIP/505-00000008'
<--- SIP read from 213.144.184.12:5060 --->
SIP/2.0 200 OK
CSeq: 104 BYE
Via: SIP/2.0/UDP 195.184.196.222:5060;branch=z9hG4bK325599a1;rport
From: "505" <sip:WMAG006 at foib.future-b.eu>;tag=as51bae2b2
Call-ID: 356e350427ca0d40294efae962c045f5 at foib.future-b.eu
To: <sip:380623880525 at foib.future-b.eu>;tag=1037221576942347
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'356e350427ca0d40294efae962c045f5 at foib.future-b.eu' Method: INVITE
Really destroying SIP dialog
'3c267b928b29-b7et3f97hqmi at snom320-0004132485B0' Method: BYE
tele2web-ukraine*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk ending (0).
Issue History
Date Modified Username Field Change
======================================================================
2010-09-21 04:48 romirikos Note Added: 0127181
======================================================================
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