[Home]

Summary:ASTERISK-12794: blindxfer doesn't work properly
Reporter:Daniel Wagner (dwagner)Labels:
Date Opened:2008-09-29 02:51:39Date Closed:2008-10-03 12:02:51
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Applications/app_transfer
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
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.

****** ADDITIONAL INFORMATION ******

   -- Executing [12@from-internal:1] Macro("SIP/13-08216948", "exten-vm|12|12") in new stack
   -- Executing [s@macro-exten-vm:1] Macro("SIP/13-08216948", "user-callerid") in new stack
   -- Executing [s@macro-user-callerid:1] NoOp("SIP/13-08216948", "user-callerid: device 13") in new stack
   -- Executing [s@macro-user-callerid:2] Set("SIP/13-08216948", "AMPUSER=13") in new stack
   -- Executing [s@macro-user-callerid:3] GotoIf("SIP/13-08216948", "0?report") in new stack
   -- Executing [s@macro-user-callerid:4] ExecIf("SIP/13-08216948", "1|Set|REALCALLERIDNUM=13") in new stack
   -- Executing [s@macro-user-callerid:5] NoOp("SIP/13-08216948", "REALCALLERIDNUM is 13") in new stack
   -- Executing [s@macro-user-callerid:6] Set("SIP/13-08216948", "AMPUSER=13") in new stack
   -- Executing [s@macro-user-callerid:7] Set("SIP/13-08216948", "AMPUSERCIDNAME=Klappe D") in new stack
   -- Executing [s@macro-user-callerid:8] GotoIf("SIP/13-08216948", "0?report") in new stack
   -- Executing [s@macro-user-callerid:9] Set("SIP/13-08216948", "AMPUSERCID=13") in new stack
   -- Executing [s@macro-user-callerid:10] Set("SIP/13-08216948", "CALLERID(all)="Klappe D" <13>") in new stack
   -- Executing [s@macro-user-callerid:11] Set("SIP/13-08216948", "REALCALLERIDNUM=13") in new stack
   -- Executing [s@macro-user-callerid:12] ExecIf("SIP/13-08216948", "0|Set|CHANNEL(language)=") in new stack
   -- Executing [s@macro-user-callerid:13] NoOp("SIP/13-08216948", "TTL:  ARG1: 12") in new stack
   -- Executing [s@macro-user-callerid:14] GotoIf("SIP/13-08216948", "0?continue") in new stack
   -- Executing [s@macro-user-callerid:15] Set("SIP/13-08216948", "__TTL=64") in new stack
   -- Executing [s@macro-user-callerid:16] GotoIf("SIP/13-08216948", "1?continue") in new stack
   -- Goto (macro-user-callerid,s,23)
   -- Executing [s@macro-user-callerid:23] NoOp("SIP/13-08216948", "Using CallerID "Klappe D" <13>") in new stack
   -- Executing [s@macro-exten-vm:2] Set("SIP/13-08216948", "RingGroupMethod=none") in new stack
   -- Executing [s@macro-exten-vm:3] Set("SIP/13-08216948", "VMBOX=12") in new stack
   -- Executing [s@macro-exten-vm:4] Set("SIP/13-08216948", "EXTTOCALL=12") in new stack
   -- Executing [s@macro-exten-vm:5] Set("SIP/13-08216948", "CFUEXT=") in new stack
   -- Executing [s@macro-exten-vm:6] Set("SIP/13-08216948", "CFBEXT=") in new stack
   -- Executing [s@macro-exten-vm:7] Set("SIP/13-08216948", "RT=35") in new stack
   -- Executing [s@macro-exten-vm:8] Macro("SIP/13-08216948", "record-enable|12|IN") in new stack
   -- Executing [s@macro-record-enable:1] GotoIf("SIP/13-08216948", "0?2:4") in new stack
   -- Goto (macro-record-enable,s,4)
   -- Executing [s@macro-record-enable:4] AGI("SIP/13-08216948", "recordingcheck|20080929-095548|1222674948.2") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck|20080929-095548|1222674948.2: Inbound recording not enabled
   -- AGI Script recordingcheck completed, returning 0
   -- Executing [s@macro-record-enable:5] NoOp("SIP/13-08216948", "No recording needed") in new stack
   -- Executing [s@macro-exten-vm:9] Macro("SIP/13-08216948", "dial|35|tr|12") in new stack
   -- Executing [s@macro-dial:1] GotoIf("SIP/13-08216948", "1?dial") in new stack
   -- Goto (macro-dial,s,3)
   -- Executing [s@macro-dial:3] AGI("SIP/13-08216948", "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 D' number is '13'
 dialparties.agi: USE_CONFIRMATION:  'FALSE'
 dialparties.agi: RINGGROUP_INDEX:   ''
 dialparties.agi: Methodology of ring is  'none'
   --  dialparties.agi: Added extension 12 to extension map
   --  dialparties.agi: Extension 12 cf is disabled
   --  dialparties.agi: Extension 12 do not disturb is disabled
      >  dialparties.agi: extnum 12 has:  cw: 0; hascfb: 0 [] hascfu: 0 []
      >  dialparties.agi: ExtensionState: 0
 dialparties.agi: Extension 12 has ExtensionState: 0
   --  dialparties.agi: Checking CW and CFB status for extension 12
   --  dialparties.agi: dbset CALLTRACE/12 to 13
   --  dialparties.agi: Filtered ARG3: 12
 == Manager 'admin' logged off from 127.0.0.1
   -- AGI Script dialparties.agi completed, returning 0
   -- Executing [s@macro-dial:7] Dial("SIP/13-08216948", "SIP/12|35|tr") in new stack
   -- Called 12
   -- SIP/12-0821b3d0 is ringing
   -- SIP/12-0821b3d0 is ringing
   -- SIP/12-0821b3d0 is ringing
   -- SIP/12-0821b3d0 answered SIP/13-08216948
   -- Started music on hold, class 'default', on SIP/13-08216948
   -- Stopped music on hold on SIP/13-08216948
   -- Executing [h@from-internal-xfer:1] Macro("SIP/13-08216948", "hangupcall") in new stack
   -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/13-08216948", "w") in new stack
   -- Executing [s@macro-hangupcall:2] NoCDR("SIP/13-08216948", "") in new stack
   -- Executing [s@macro-hangupcall:3] GotoIf("SIP/13-08216948", "1?skiprg") in new stack
   -- Goto (macro-hangupcall,s,6)
   -- Executing [s@macro-hangupcall:6] GotoIf("SIP/13-08216948", "1?skipblkvm") in new stack
   -- Goto (macro-hangupcall,s,9)
   -- Executing [s@macro-hangupcall:9] GotoIf("SIP/13-08216948", "1?theend") in new stack
   -- Goto (macro-hangupcall,s,11)
   -- Executing [s@macro-hangupcall:11] Hangup("SIP/13-08216948", "") in new stack
 == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/13-08216948' in macro 'hangupcall'
 == Spawn h extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/13-08216948'
 == Spawn extension (macro-hangupcall, 14, 0) exited non-zero on 'SIP/13-08216948' in macro 'dial'
 == Spawn extension (macro-hangupcall, 14, 0) exited non-zero on 'SIP/13-08216948' in macro 'exten-vm'
 == Spawn extension (macro-hangupcall, 14, 0) exited non-zero on 'SIP/13-08216948'
Comments:By: Daniel Wagner (dwagner) 2008-09-29 07:59:52

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@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@210.0.0.227' Method: OPTIONS
Really destroying SIP dialog '6024f3d02d4d2d735473a9ec7bc05472@127.0.0.1' Method: REGISTER

<--- SIP read from 210.0.0.151:2063 --->
INVITE sip:13@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@x.x.x.x>;tag=9u035nymzo

To: <sip:13@x.x.x.x;user=phone>

Call-ID: 3c2685145cc6-8rext4k3c3j5@snom360-000413237AF3

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:12@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@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@x.x.x.x>;tag=9u035nymzo

To: <sip:13@x.x.x.x;user=phone>;tag=as660670d1

Call-ID: 3c2685145cc6-8rext4k3c3j5@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@snom360-000413237AF3' in 32000 ms (Method: INVITE)
Found user '12'

<--- SIP read from 210.0.0.151:2063 --->
ACK sip:13@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@x.x.x.x>;tag=9u035nymzo

To: <sip:13@x.x.x.x;user=phone>;tag=as660670d1

Call-ID: 3c2685145cc6-8rext4k3c3j5@snom360-000413237AF3

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:12@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@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@x.x.x.x>;tag=9u035nymzo

To: <sip:13@x.x.x.x;user=phone>

Call-ID: 3c2685145cc6-8rext4k3c3j5@snom360-000413237AF3

CSeq: 2 INVITE

Max-Forwards: 70

Contact: <sip:12@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@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@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@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@x.x.x.x>;tag=9u035nymzo

To: <sip:13@x.x.x.x;user=phone>

Call-ID: 3c2685145cc6-8rext4k3c3j5@snom360-000413237AF3

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:13@x.x.x.x>

Content-Length: 0




<------------>
   -- Executing [13@from-internal:1] Macro("SIP/12-082134b0", "exten-vm|13|13") in new stack
   -- Executing [s@macro-exten-vm:1] Macro("SIP/12-082134b0", "user-callerid") in new stack
   -- Executing [s@macro-user-callerid:1] NoOp("SIP/12-082134b0", "user-callerid: device 12") in new stack
   -- Executing [s@macro-user-callerid:2] Set("SIP/12-082134b0", "AMPUSER=12") in new stack
   -- Executing [s@macro-user-callerid:3] GotoIf("SIP/12-082134b0", "0?report") in new stack
   -- Executing [s@macro-user-callerid:4] ExecIf("SIP/12-082134b0", "1|Set|REALCALLERIDNUM=12") in new stack
   -- Executing [s@macro-user-callerid:5] NoOp("SIP/12-082134b0", "REALCALLERIDNUM is 12") in new stack
   -- Executing [s@macro-user-callerid:6] Set("SIP/12-082134b0", "AMPUSER=12") in new stack
   -- Executing [s@macro-user-callerid:7] Set("SIP/12-082134b0", "AMPUSERCIDNAME=Klappe C") in new stack
   -- Executing [s@macro-user-callerid:8] GotoIf("SIP/12-082134b0", "0?report") in new stack
   -- Executing [s@macro-user-callerid:9] Set("SIP/12-082134b0", "AMPUSERCID=12") in new stack
   -- Executing [s@macro-user-callerid:10] Set("SIP/12-082134b0", "CALLERID(all)="Klappe C" <12>") in new stack
   -- Executing [s@macro-user-callerid:11] Set("SIP/12-082134b0", "REALCALLERIDNUM=12") in new stack
   -- Executing [s@macro-user-callerid:12] ExecIf("SIP/12-082134b0", "0|Set|CHANNEL(language)=") in new stack
   -- Executing [s@macro-user-callerid:13] NoOp("SIP/12-082134b0", "TTL:  ARG1: 13") in new stack
   -- Executing [s@macro-user-callerid:14] GotoIf("SIP/12-082134b0", "0?continue") in new stack
   -- Executing [s@macro-user-callerid:15] Set("SIP/12-082134b0", "__TTL=64") in new stack
   -- Executing [s@macro-user-callerid:16] GotoIf("SIP/12-082134b0", "1?continue") in new stack
   -- Goto (macro-user-callerid,s,23)
   -- Executing [s@macro-user-callerid:23] NoOp("SIP/12-082134b0", "Using CallerID "Klappe C" <12>") in new stack
   -- Executing [s@macro-exten-vm:2] Set("SIP/12-082134b0", "RingGroupMethod=none") in new stack
   -- Executing [s@macro-exten-vm:3] Set("SIP/12-082134b0", "VMBOX=13") in new stack
   -- Executing [s@macro-exten-vm:4] Set("SIP/12-082134b0", "EXTTOCALL=13") in new stack
   -- Executing [s@macro-exten-vm:5] Set("SIP/12-082134b0", "CFUEXT=") in new stack
   -- Executing [s@macro-exten-vm:6] Set("SIP/12-082134b0", "CFBEXT=") in new stack
   -- Executing [s@macro-exten-vm:7] Set("SIP/12-082134b0", "RT=35") in new stack
   -- Executing [s@macro-exten-vm:8] Macro("SIP/12-082134b0", "record-enable|13|IN") in new stack
   -- Executing [s@macro-record-enable:1] GotoIf("SIP/12-082134b0", "0?2:4") in new stack
   -- Goto (macro-record-enable,s,4)
   -- Executing [s@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@macro-record-enable:5] NoOp("SIP/12-082134b0", "No recording needed") in new stack
   -- Executing [s@macro-exten-vm:9] Macro("SIP/12-082134b0", "dial|35|tr|13") in new stack
   -- Executing [s@macro-dial:1] GotoIf("SIP/12-082134b0", "1?dial") in new stack
   -- Goto (macro-dial,s,3)
   -- Executing [s@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@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@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>

Contact: <sip:12@x.x.x.x>

Call-ID: 432991441a9e32f655ec6b4c799c8222@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@x.x.x.x>;tag=9u035nymzo

To: <sip:13@x.x.x.x;user=phone>;tag=as794e6311

Call-ID: 3c2685145cc6-8rext4k3c3j5@snom360-000413237AF3

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:13@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

Call-ID: 432991441a9e32f655ec6b4c799c8222@x.x.x.x

CSeq: 102 INVITE

Contact: <sip:13@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

Call-ID: 432991441a9e32f655ec6b4c799c8222@x.x.x.x

CSeq: 102 INVITE

Contact: <sip:13@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

Call-ID: 432991441a9e32f655ec6b4c799c8222@x.x.x.x

CSeq: 102 INVITE

Contact: <sip:13@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@210.0.0.167:2060;line=4gl2bzgn>
set_destination: Parsing <sip:13@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@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

Contact: <sip:12@x.x.x.x>

Call-ID: 432991441a9e32f655ec6b4c799c8222@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@x.x.x.x>;tag=9u035nymzo

To: <sip:13@x.x.x.x;user=phone>;tag=as794e6311

Call-ID: 3c2685145cc6-8rext4k3c3j5@snom360-000413237AF3

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:13@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@x.x.x.x SIP/2.0

Via: SIP/2.0/UDP 210.0.0.151:2063;branch=z9hG4bK-omoy1csi3tza;rport

From: <sip:12@x.x.x.x>;tag=9u035nymzo

To: <sip:13@x.x.x.x;user=phone>;tag=as794e6311

Call-ID: 3c2685145cc6-8rext4k3c3j5@snom360-000413237AF3

CSeq: 2 ACK

Max-Forwards: 70

Contact: <sip:12@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@x.x.x.x SIP/2.0

Via: SIP/2.0/UDP 210.0.0.167:2060;branch=z9hG4bK-rfbflzx7cqpz;rport

From: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

To: "Klappe C" <sip:12@x.x.x.x>;tag=as1664a8af

Call-ID: 432991441a9e32f655ec6b4c799c8222@x.x.x.x

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:13@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@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

To: "Klappe C" <sip:12@x.x.x.x>;tag=as1664a8af

Call-ID: 432991441a9e32f655ec6b4c799c8222@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@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@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

To: "Klappe C" <sip:12@x.x.x.x>;tag=as1664a8af

Call-ID: 432991441a9e32f655ec6b4c799c8222@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@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@x.x.x.x SIP/2.0

Via: SIP/2.0/UDP 210.0.0.167:2060;branch=z9hG4bK-j8aofetphhno;rport

From: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

To: "Klappe C" <sip:12@x.x.x.x>;tag=as1664a8af

Call-ID: 432991441a9e32f655ec6b4c799c8222@x.x.x.x

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:13@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@x.x.x.x SIP/2.0

Via: SIP/2.0/UDP 210.0.0.167:2060;branch=z9hG4bK-wuqgqiuhy5gj;rport

From: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

To: "Klappe C" <sip:12@x.x.x.x>;tag=as1664a8af

Call-ID: 432991441a9e32f655ec6b4c799c8222@x.x.x.x

CSeq: 2 REFER

Max-Forwards: 70

Contact: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;flow-id=1

Refer-To: sip:14@x.x.x.x;user=phone

Referred-By: sip:13@x.x.x.x

User-Agent: snom320/6.5.16

Content-Length: 0




<------------->
--- (12 headers 0 lines) ---
Call 432991441a9e32f655ec6b4c799c8222@x.x.x.x got a SIP call transfer from caller: (REFER)!
SIP transfer to extension 14@from-internal-xfer by 13@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@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

To: "Klappe C" <sip:12@x.x.x.x>;tag=as1664a8af

Call-ID: 432991441a9e32f655ec6b4c799c8222@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@x.x.x.x>

Content-Length: 0




<------------>
set_destination: Parsing <sip:13@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@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

Contact: <sip:12@x.x.x.x>

Call-ID: 432991441a9e32f655ec6b4c799c8222@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@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@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

Contact: <sip:12@x.x.x.x>

Call-ID: 432991441a9e32f655ec6b4c799c8222@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@from-internal-xfer:1] Macro("SIP/12-082134b0", "hangupcall") in new stack
   -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/12-082134b0", "w") in new stack
   -- Executing [s@macro-hangupcall:2] NoCDR("SIP/12-082134b0", "") in new stack
   -- Executing [s@macro-hangupcall:3] GotoIf("SIP/12-082134b0", "1?skiprg") in new stack
   -- Goto (macro-hangupcall,s,6)
   -- Executing [s@macro-hangupcall:6] GotoIf("SIP/12-082134b0", "1?skipblkvm") in new stack
   -- Goto (macro-hangupcall,s,9)
   -- Executing [s@macro-hangupcall:9] GotoIf("SIP/12-082134b0", "1?theend") in new stack
   -- Goto (macro-hangupcall,s,11)
   -- Executing [s@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@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@snom360-000413237AF3' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:12@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@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@x.x.x.x;user=phone>;tag=as794e6311

To: <sip:12@x.x.x.x>;tag=9u035nymzo

Call-ID: 3c2685145cc6-8rext4k3c3j5@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

Call-ID: 432991441a9e32f655ec6b4c799c8222@x.x.x.x

CSeq: 103 NOTIFY

Contact: <sip:13@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

Call-ID: 432991441a9e32f655ec6b4c799c8222@x.x.x.x

CSeq: 104 NOTIFY

Contact: <sip:13@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 #1 (NAT) to 210.0.0.167:2060:
NOTIFY sip:13@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

Contact: <sip:12@x.x.x.x>

Call-ID: 432991441a9e32f655ec6b4c799c8222@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@x.x.x.x SIP/2.0

Via: SIP/2.0/UDP 210.0.0.167:2060;branch=z9hG4bK-pda13kknsuvs;rport

From: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

To: "Klappe C" <sip:12@x.x.x.x>;tag=as1664a8af

Call-ID: 432991441a9e32f655ec6b4c799c8222@x.x.x.x

CSeq: 3 BYE

Max-Forwards: 70

Contact: <sip:13@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@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@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

To: "Klappe C" <sip:12@x.x.x.x>;tag=as1664a8af

Call-ID: 432991441a9e32f655ec6b4c799c8222@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@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@x.x.x.x>;tag=as1664a8af

To: <sip:13@210.0.0.167:2060;line=4gl2bzgn>;tag=9dihwbt9u0

Call-ID: 432991441a9e32f655ec6b4c799c8222@x.x.x.x

CSeq: 103 NOTIFY

Contact: <sip:13@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@x.x.x.x;user=phone>;tag=as794e6311

To: <sip:12@x.x.x.x>;tag=9u035nymzo

Call-ID: 3c2685145cc6-8rext4k3c3j5@snom360-000413237AF3

CSeq: 102 BYE

Contact: <sip:12@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@snom360-000413237AF3' Method: ACK

ipefon097*CLI> exit
Really destroying SIP dialog '432991441a9e32f655ec6b4c799c8222@x.x.x.x' Method: BYE

ipefon097*CLI> exit

Executing last minute cleanups
By: Daniel Wagner (dwagner) 2008-09-29 09:52:13

move this issue to 0013584

found the bug!
By: Digium Subversion (svnbot) 2008-10-03 12:02:49

Repository: asterisk
Revision: 146026

U   branches/1.4/res/res_features.c

------------------------------------------------------------------------
r146026 | murf | 2008-10-03 12:02:48 -0500 (Fri, 03 Oct 2008) | 18 lines

(closes issue ASTERISK-12794)
Reported by: dwagner

(closes issue ASTERISK-12798)
Reported by: dwagner
Tested by: murf, putnopvut

The thought occurred to me that the res= from the extension spawn
was ending up being returned from the bridge.

"Thou shalt not poison the return value". Made the change
and it appears to allow blind xfers to work as normal.

If I'm wrong, reopen the bugs. But it looks good to me!

Many thanks to putnopvut for helping me reproduce this!


------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=146026