[asterisk-bugs] [Asterisk 0013579]: blindxfer doesn't work properly
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Sep 29 07:59:53 CDT 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=13579
======================================================================
Reported By: dwagner
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 13579
Category: Applications/app_transfer
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 144925M
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 2008-09-29 02:51 CDT
Last Modified: 2008-09-29 07:59 CDT
======================================================================
Summary: blindxfer doesn't work properly
Description:
i updated to the latest svn r144925M. we have the behavour, that a blind
call transfer cause a call drop. in the latest stable version 1.4.22-rc5 it
works properly. i also see that if i don't answer the call and transfer it
directly to another extension it works perfect.
======================================================================
----------------------------------------------------------------------
(0092942) dwagner (reporter) - 2008-09-29 07:59
http://bugs.digium.com/view.php?id=13579#c92942
----------------------------------------------------------------------
here is the sip debug
call 12 -> 13, 13 answer, transfer 14, hangup
Asterisk SVN-branch-1.4-r144925, Copyright (C) 1999 - 2008 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.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
Connected to Asterisk SVN-branch-1.4-r144925 currently running on
ipefon097 (pid = 28405)Verbosity is at least 6
ipefon097*CLI> sip debug
SIP Debugging enabled
The 'sip debug' command is deprecated and will be removed in a future
release. Please use 'sip set debug' instead.
Really destroying SIP dialog
'03e9e52127e034870e5103af66e71f24 at 210.0.0.227' Method: OPTIONS
Really destroying SIP dialog '6024f3d02d4d2d735473a9ec7bc05472 at 127.0.0.1'
Method: REGISTER
<--- SIP read from 210.0.0.151:2063 --->
INVITE sip:13 at x.x.x.x;user=phone SIP/2.0
Via: SIP/2.0/UDP 210.0.0.151:2063;branch=z9hG4bK-axr1gnro9h3z;rport
From: <sip:12 at x.x.x.x>;tag=9u035nymzo
To: <sip:13 at x.x.x.x;user=phone>
Call-ID: 3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:12 at 210.0.0.151:2063;line=4u4y9gvi>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.5.13
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
Content-Type: application/sdp
Content-Length: 446
v=0
o=root 1743711491 1743711491 IN IP4 210.0.0.151
s=call
c=IN IP4 210.0.0.151
t=0 0
m=audio 61050 RTP/AVP 8 9 0 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:++wyHTPLfpL9WuFkXr7G1Bz7X4ebp6OB9S1Xisr5
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:0 pcmu/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 18 lines) ---
Sending to 210.0.0.151 : 2063 (NAT)
Using INVITE request as basis request -
3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
<--- Reliably Transmitting (NAT) to 210.0.0.151:2063 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
210.0.0.151:2063;branch=z9hG4bK-axr1gnro9h3z;received=210.0.0.151;rport=2063
From: <sip:12 at x.x.x.x>;tag=9u035nymzo
To: <sip:13 at x.x.x.x;user=phone>;tag=as660670d1
Call-ID: 3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="7b55a9fd"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3' in 32000 ms (Method:
INVITE)
Found user '12'
<--- SIP read from 210.0.0.151:2063 --->
ACK sip:13 at x.x.x.x;user=phone SIP/2.0
Via: SIP/2.0/UDP 210.0.0.151:2063;branch=z9hG4bK-axr1gnro9h3z;rport
From: <sip:12 at x.x.x.x>;tag=9u035nymzo
To: <sip:13 at x.x.x.x;user=phone>;tag=as660670d1
Call-ID: 3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:12 at 210.0.0.151:2063;line=4u4y9gvi>;flow-id=1
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from 210.0.0.151:2063 --->
INVITE sip:13 at x.x.x.x;user=phone SIP/2.0
Via: SIP/2.0/UDP 210.0.0.151:2063;branch=z9hG4bK-ij47tulp7ubn;rport
From: <sip:12 at x.x.x.x>;tag=9u035nymzo
To: <sip:13 at x.x.x.x;user=phone>
Call-ID: 3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:12 at 210.0.0.151:2063;line=4u4y9gvi>;flow-id=1
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/6.5.13
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="12",realm="asterisk",nonce="7b55a9fd",uri="sip:13 at x.x.x.x;user=phone",response="693c0fb1fe76af7e56dabf46afbeb29b",algorithm=MD5
Content-Type: application/sdp
Content-Length: 446
v=0
o=root 1743711491 1743711491 IN IP4 210.0.0.151
s=call
c=IN IP4 210.0.0.151
t=0 0
m=audio 61050 RTP/AVP 8 9 0 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:++wyHTPLfpL9WuFkXr7G1Bz7X4ebp6OB9S1Xisr5
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:0 pcmu/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 18 lines) ---
Sending to 210.0.0.151 : 2063 (NAT)
Using INVITE request as basis request -
3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
Found user '12'
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 210.0.0.151:61050
Found audio description format pcma for ID 8
Found audio description format g722 for ID 9
Found audio description format pcmu for ID 0
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 - 0xc (ulaw|alaw), peer - audio=0x110f
(g723|gsm|ulaw|alaw|g729|g722)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 210.0.0.151:61050
Looking for 13 in from-internal (domain x.x.x.x)
list_route: hop: <sip:12 at 210.0.0.151:2063;line=4u4y9gvi>
<--- Transmitting (NAT) to 210.0.0.151:2063 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
210.0.0.151:2063;branch=z9hG4bK-ij47tulp7ubn;received=210.0.0.151;rport=2063
From: <sip:12 at x.x.x.x>;tag=9u035nymzo
To: <sip:13 at x.x.x.x;user=phone>
Call-ID: 3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13 at x.x.x.x>
Content-Length: 0
<------------>
-- Executing [13 at from-internal:1] Macro("SIP/12-082134b0",
"exten-vm|13|13") in new stack
-- Executing [s at macro-exten-vm:1] Macro("SIP/12-082134b0",
"user-callerid") in new stack
-- Executing [s at macro-user-callerid:1] NoOp("SIP/12-082134b0",
"user-callerid: device 12") in new stack
-- Executing [s at macro-user-callerid:2] Set("SIP/12-082134b0",
"AMPUSER=12") in new stack
-- Executing [s at macro-user-callerid:3] GotoIf("SIP/12-082134b0",
"0?report") in new stack
-- Executing [s at macro-user-callerid:4] ExecIf("SIP/12-082134b0",
"1|Set|REALCALLERIDNUM=12") in new stack
-- Executing [s at macro-user-callerid:5] NoOp("SIP/12-082134b0",
"REALCALLERIDNUM is 12") in new stack
-- Executing [s at macro-user-callerid:6] Set("SIP/12-082134b0",
"AMPUSER=12") in new stack
-- Executing [s at macro-user-callerid:7] Set("SIP/12-082134b0",
"AMPUSERCIDNAME=Klappe C") in new stack
-- Executing [s at macro-user-callerid:8] GotoIf("SIP/12-082134b0",
"0?report") in new stack
-- Executing [s at macro-user-callerid:9] Set("SIP/12-082134b0",
"AMPUSERCID=12") in new stack
-- Executing [s at macro-user-callerid:10] Set("SIP/12-082134b0",
"CALLERID(all)="Klappe C" <12>") in new stack
-- Executing [s at macro-user-callerid:11] Set("SIP/12-082134b0",
"REALCALLERIDNUM=12") in new stack
-- Executing [s at macro-user-callerid:12] ExecIf("SIP/12-082134b0",
"0|Set|CHANNEL(language)=") in new stack
-- Executing [s at macro-user-callerid:13] NoOp("SIP/12-082134b0", "TTL:
ARG1: 13") in new stack
-- Executing [s at macro-user-callerid:14] GotoIf("SIP/12-082134b0",
"0?continue") in new stack
-- Executing [s at macro-user-callerid:15] Set("SIP/12-082134b0",
"__TTL=64") in new stack
-- Executing [s at macro-user-callerid:16] GotoIf("SIP/12-082134b0",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s at macro-user-callerid:23] NoOp("SIP/12-082134b0", "Using
CallerID "Klappe C" <12>") in new stack
-- Executing [s at macro-exten-vm:2] Set("SIP/12-082134b0",
"RingGroupMethod=none") in new stack
-- Executing [s at macro-exten-vm:3] Set("SIP/12-082134b0", "VMBOX=13")
in new stack
-- Executing [s at macro-exten-vm:4] Set("SIP/12-082134b0",
"EXTTOCALL=13") in new stack
-- Executing [s at macro-exten-vm:5] Set("SIP/12-082134b0", "CFUEXT=") in
new stack
-- Executing [s at macro-exten-vm:6] Set("SIP/12-082134b0", "CFBEXT=") in
new stack
-- Executing [s at macro-exten-vm:7] Set("SIP/12-082134b0", "RT=35") in
new stack
-- Executing [s at macro-exten-vm:8] Macro("SIP/12-082134b0",
"record-enable|13|IN") in new stack
-- Executing [s at macro-record-enable:1] GotoIf("SIP/12-082134b0",
"0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s at macro-record-enable:4] AGI("SIP/12-082134b0",
"recordingcheck|20080929-145634|1222692994.2") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20080929-145634|1222692994.2: Inbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s at macro-record-enable:5] NoOp("SIP/12-082134b0", "No
recording needed") in new stack
-- Executing [s at macro-exten-vm:9] Macro("SIP/12-082134b0",
"dial|35|tr|13") in new stack
-- Executing [s at macro-dial:1] GotoIf("SIP/12-082134b0", "1?dial") in
new stack
-- Goto (macro-dial,s,3)
-- Executing [s at macro-dial:3] AGI("SIP/12-082134b0",
"dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'Klappe C' number is '12'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 13 to extension map
-- dialparties.agi: Extension 13 cf is disabled
-- dialparties.agi: Extension 13 do not disturb is disabled
> dialparties.agi: extnum 13 has: cw: 0; hascfb: 0 [] hascfu: 0
[]
> dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 13 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 13
-- dialparties.agi: dbset CALLTRACE/13 to 12
-- dialparties.agi: Filtered ARG3: 13
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s at macro-dial:7] Dial("SIP/12-082134b0", "SIP/13|35|tr")
in new stack
Audio is at x.x.x.x port 11196
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 210.0.0.167:2060:
INVITE sip:13 at 210.0.0.167:2060;line=4gl2bzgn SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK708cf93f;rport
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>
Contact: <sip:12 at x.x.x.x>
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 29 Sep 2008 12:56:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 28405 28405 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 11196 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 13
<--- Transmitting (NAT) to 210.0.0.151:2063 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
210.0.0.151:2063;branch=z9hG4bK-ij47tulp7ubn;received=210.0.0.151;rport=2063
From: <sip:12 at x.x.x.x>;tag=9u035nymzo
To: <sip:13 at x.x.x.x;user=phone>;tag=as794e6311
Call-ID: 3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13 at x.x.x.x>
Content-Length: 0
<------------>
<--- SIP read from 210.0.0.167:2060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK708cf93f;rport=5060
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 102 INVITE
Contact: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- SIP/13-08219348 is ringing
<--- SIP read from 210.0.0.167:2060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK708cf93f;rport=5060
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 102 INVITE
Contact: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;flow-id=1
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- SIP/13-08219348 is ringing
<--- SIP read from 210.0.0.167:2060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK708cf93f;rport=5060
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 102 INVITE
Contact: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;flow-id=1
User-Agent: snom320/6.5.16
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Content-Type: application/sdp
Content-Length: 347
v=0
o=root 483315004 483315005 IN IP4 210.0.0.167
s=call
c=IN IP4 210.0.0.167
t=0 0
m=audio 62328 RTP/AVP 0 8 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:GTtZrKbR9x9lKp+axKy383JrL+tbSyQ++/+ppkSe
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=encryption:optional
a=sendrecv
<------------->
--- (13 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 210.0.0.167:62328
Found audio description format pcmu for ID 0
Found audio description format pcma for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0
(nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 210.0.0.167:62328
list_route: hop: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>
set_destination: Parsing <sip:13 at 210.0.0.167:2060;line=4gl2bzgn> for
address/port to send to
set_destination: set destination to 210.0.0.167, port 2060
Transmitting (NAT) to 210.0.0.167:2060:
ACK sip:13 at 210.0.0.167:2060;line=4gl2bzgn SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK67520ae3;rport
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
Contact: <sip:12 at x.x.x.x>
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/13-08219348 answered SIP/12-082134b0
Audio is at x.x.x.x port 18094
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 210.0.0.151:2063 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
210.0.0.151:2063;branch=z9hG4bK-ij47tulp7ubn;received=210.0.0.151;rport=2063
From: <sip:12 at x.x.x.x>;tag=9u035nymzo
To: <sip:13 at x.x.x.x;user=phone>;tag=as794e6311
Call-ID: 3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:13 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 28405 28405 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 18094 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from 210.0.0.151:2063 --->
ACK sip:13 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 210.0.0.151:2063;branch=z9hG4bK-omoy1csi3tza;rport
From: <sip:12 at x.x.x.x>;tag=9u035nymzo
To: <sip:13 at x.x.x.x;user=phone>;tag=as794e6311
Call-ID: 3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:12 at 210.0.0.151:2063;line=4u4y9gvi>;flow-id=1
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from 210.0.0.167:2060 --->
INVITE sip:12 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 210.0.0.167:2060;branch=z9hG4bK-rfbflzx7cqpz;rport
From: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
To: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/6.5.16
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
Content-Type: application/sdp
Content-Length: 473
v=0
o=root 483315004 483315006 IN IP4 210.0.0.167
s=call
c=IN IP4 210.0.0.167
t=0 0
m=audio 62328 RTP/AVP 0 8 9 98 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:GTtZrKbR9x9lKp+axKy383JrL+tbSyQ++/+ppkSe
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:98 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=sendonly
<------------->
--- (18 headers 19 lines) ---
Sending to 210.0.0.167 : 2060 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 98
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 210.0.0.167:62328
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 98
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 - 0xc (ulaw|alaw), peer - audio=0x190f
(g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 210.0.0.167:62328
<--- Transmitting (NAT) to 210.0.0.167:2060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
210.0.0.167:2060;branch=z9hG4bK-rfbflzx7cqpz;received=210.0.0.167;rport=2060
From: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
To: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12 at x.x.x.x>
Content-Length: 0
<------------>
Audio is at x.x.x.x port 11196
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 210.0.0.167:2060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
210.0.0.167:2060;branch=z9hG4bK-rfbflzx7cqpz;received=210.0.0.167;rport=2060
From: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
To: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12 at x.x.x.x>
Content-Type: application/sdp
Content-Length: 270
v=0
o=root 28405 28406 IN IP4 x.x.x.x
s=session
c=IN IP4 x.x.x.x
t=0 0
m=audio 11196 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=recvonly
<------------>
-- Started music on hold, class 'default', on SIP/12-082134b0
<--- SIP read from 210.0.0.167:2060 --->
ACK sip:12 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 210.0.0.167:2060;branch=z9hG4bK-j8aofetphhno;rport
From: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
To: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;flow-id=1
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from 210.0.0.167:2060 --->
REFER sip:12 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 210.0.0.167:2060;branch=z9hG4bK-wuqgqiuhy5gj;rport
From: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
To: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 2 REFER
Max-Forwards: 70
Contact: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;flow-id=1
Refer-To: sip:14 at x.x.x.x;user=phone
Referred-By: sip:13 at x.x.x.x
User-Agent: snom320/6.5.16
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Call 432991441a9e32f655ec6b4c799c8222 at x.x.x.x got a SIP call transfer from
caller: (REFER)!
SIP transfer to extension 14 at from-internal-xfer by 13 at x.x.x.x
<--- Transmitting (NAT) to 210.0.0.167:2060 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP
210.0.0.167:2060;branch=z9hG4bK-wuqgqiuhy5gj;received=210.0.0.167;rport=2060
From: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
To: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12 at x.x.x.x>
Content-Length: 0
<------------>
set_destination: Parsing <sip:13 at 210.0.0.167:2060;line=4gl2bzgn> for
address/port to send to
set_destination: set destination to 210.0.0.167, port 2060
Reliably Transmitting (NAT) to 210.0.0.167:2060:
NOTIFY sip:13 at 210.0.0.167:2060;line=4gl2bzgn SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK188db2b0;rport
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
Contact: <sip:12 at x.x.x.x>
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=2
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 21
SIP/2.0 183 Ringing
---
set_destination: Parsing <sip:13 at 210.0.0.167:2060;line=4gl2bzgn> for
address/port to send to
set_destination: set destination to 210.0.0.167, port 2060
Reliably Transmitting (NAT) to 210.0.0.167:2060:
NOTIFY sip:13 at 210.0.0.167:2060;line=4gl2bzgn SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK33da0238;rport
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
Contact: <sip:12 at x.x.x.x>
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=2
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 16
SIP/2.0 200 Ok
---
-- Stopped music on hold on SIP/12-082134b0
-- Executing [h at from-internal-xfer:1] Macro("SIP/12-082134b0",
"hangupcall") in new stack
-- Executing [s at macro-hangupcall:1] ResetCDR("SIP/12-082134b0", "w")
in new stack
-- Executing [s at macro-hangupcall:2] NoCDR("SIP/12-082134b0", "") in
new stack
-- Executing [s at macro-hangupcall:3] GotoIf("SIP/12-082134b0",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s at macro-hangupcall:6] GotoIf("SIP/12-082134b0",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] GotoIf("SIP/12-082134b0",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s at macro-hangupcall:11] Hangup("SIP/12-082134b0", "") in
new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/12-082134b0' in macro 'hangupcall'
== Spawn h extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/12-082134b0'
Scheduling destruction of SIP dialog
'432991441a9e32f655ec6b4c799c8222 at x.x.x.x' in 6400 ms (Method: REFER)
== Spawn extension (macro-hangupcall, 14, 0) exited non-zero on
'SIP/12-082134b0' in macro 'dial'
== Spawn extension (macro-hangupcall, 14, 0) exited non-zero on
'SIP/12-082134b0' in macro 'exten-vm'
== Spawn extension (macro-hangupcall, 14, 0) exited non-zero on
'SIP/12-082134b0'
Scheduling destruction of SIP dialog
'3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:12 at 210.0.0.151:2063;line=4u4y9gvi> for
address/port to send to
set_destination: set destination to 210.0.0.151, port 2063
Reliably Transmitting (NAT) to 210.0.0.151:2063:
BYE sip:12 at 210.0.0.151:2063;line=4u4y9gvi SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK1ce4f442;rport
From: <sip:13 at x.x.x.x;user=phone>;tag=as794e6311
To: <sip:12 at x.x.x.x>;tag=9u035nymzo
Call-ID: 3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<--- SIP read from 210.0.0.167:2060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK188db2b0;rport=5060
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 103 NOTIFY
Contact: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;flow-id=1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 210.0.0.167:2060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK33da0238;rport=5060
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 104 NOTIFY
Contact: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;flow-id=1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
Retransmitting http://bugs.digium.com/view.php?id=1 (NAT) to 210.0.0.167:2060:
NOTIFY sip:13 at 210.0.0.167:2060;line=4gl2bzgn SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK188db2b0;rport
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
Contact: <sip:12 at x.x.x.x>
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=2
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 21
SIP/2.0 183 Ringing
---
<--- SIP read from 210.0.0.167:2060 --->
BYE sip:12 at x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 210.0.0.167:2060;branch=z9hG4bK-pda13kknsuvs;rport
From: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
To: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 3 BYE
Max-Forwards: 70
Contact: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;flow-id=1
User-Agent: snom320/6.5.16
RTP-RxStat:
Total_Rx_Pkts=33,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=38,Tx_Pkts=0,Remote_Tx_Pkts=0
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 210.0.0.167 : 2060 (NAT)
Scheduling destruction of SIP dialog
'432991441a9e32f655ec6b4c799c8222 at x.x.x.x' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 210.0.0.167:2060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
210.0.0.167:2060;branch=z9hG4bK-pda13kknsuvs;received=210.0.0.167;rport=2060
From: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
To: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12 at x.x.x.x>
Content-Length: 0
<------------>
<--- SIP read from 210.0.0.167:2060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK188db2b0;rport=5060
From: "Klappe C" <sip:12 at x.x.x.x>;tag=as1664a8af
To: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0
Call-ID: 432991441a9e32f655ec6b4c799c8222 at x.x.x.x
CSeq: 103 NOTIFY
Contact: <sip:13 at 210.0.0.167:2060;line=4gl2bzgn>;flow-id=1
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 210.0.0.151:2063 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK1ce4f442;rport=5060
From: <sip:13 at x.x.x.x;user=phone>;tag=as794e6311
To: <sip:12 at x.x.x.x>;tag=9u035nymzo
Call-ID: 3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3
CSeq: 102 BYE
Contact: <sip:12 at 210.0.0.151:2063;line=4u4y9gvi>;flow-id=1
User-Agent: snom360/6.5.13
RTP-RxStat:
Total_Rx_Pkts=151,Rx_Pkts=151,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=150,Tx_Pkts=150,Remote_Tx_Pkts=0
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog
'3c2685145cc6-8rext4k3c3j5 at snom360-000413237AF3' Method: ACK
ipefon097*CLI> exit
Really destroying SIP dialog '432991441a9e32f655ec6b4c799c8222 at x.x.x.x'
Method: BYE
ipefon097*CLI> exit
Executing last minute cleanups
Issue History
Date Modified Username Field Change
======================================================================
2008-09-29 07:59 dwagner Note Added: 0092942
======================================================================
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