[asterisk-bugs] [Asterisk 0016108]: Application Extenspy
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Oct 21 09:35:53 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=16108
======================================================================
Reported By: elsto
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 16108
Category: Applications/app_chanspy
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.6.1.6
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-10-21 07:15 CDT
Last Modified: 2009-10-21 09:35 CDT
======================================================================
Summary: Application Extenspy
Description:
The whisper mode isn't working properly.
I can hear the spied-on channels talking, but cannot whisper to it.
I have the same issue the option B (barge-in) as wel as option w
regards
======================================================================
----------------------------------------------------------------------
(0112523) elsto (reporter) - 2009-10-21 09:35
https://issues.asterisk.org/view.php?id=16108#c112523
----------------------------------------------------------------------
Dialplan:
exten => _*71XXX,1,Set(${EXTEN:2})
exten => _*71XXX,2,ExtenSpy(${EXTEN},w)
I use *7 to start the function and then remove *7 again with the EXTEN
variable.
Console output;
== Spying on channel SIP/1203-018e21f8
[Oct 21 16:27:34] NOTICE[16193]: app_chanspy.c:292 start_spying: Attaching
SIP/1201-0802f8d8 to SIP/1203-018e21f8
[Oct 21 16:27:34] NOTICE[16193]: app_chanspy.c:292 start_spying: Attaching
SIP/1201-0802f8d8 to SIP/1203-018e21f8
[Oct 21 16:27:34] NOTICE[16193]: app_chanspy.c:292 start_spying: Attaching
SIP/1201-0802f8d8 to SIP/1151-01a06b58
Sip debug output on peer
<--- SIP read from UDP://192.168.1.182:5060 --->
INVITE sip:*71203 at 192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.182:5060;branch=z9hG4bKc54f3d70528af1acd.fd80ab152c0418d30
Max-Forwards: 70
From: "T1201 Receptie" <sip:1201 at 192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203 at 192.168.1.18:5060>
Call-ID: dea0949d25f17307
CSeq: 28836 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "T1201 Receptie"
<sip:1201 at 192.168.1.182:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1A4BFE>"
Supported: gruu, timer, 100rel, replaces
User-Agent: Aastra 53i/2.4.1.37
Content-Type: application/sdp
Content-Length: 285
v=0
o=MxSIP 0 0 IN IP4 192.168.1.182
s=SIP Call
c=IN IP4 192.168.1.182
t=0 0
m=audio 3000 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 14 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 192.168.1.182 : 5060 (no NAT)
Using INVITE request as basis request - dea0949d25f17307
Found peer '1201' for '1201' from 192.168.1.182:5060
elspbx*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.182:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.182:5060;branch=z9hG4bKc54f3d70528af1acd.fd80ab152c0418d30;received=192.168.1.182
From: "T1201 Receptie" <sip:1201 at 192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203 at 192.168.1.18:5060>;tag=as32c804c2
Call-ID: dea0949d25f17307
CSeq: 28836 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="38034d5d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'dea0949d25f17307' in 32000 ms
(Method: INVITE)
elspbx*CLI>
<--- SIP read from UDP://192.168.1.182:5060 --->
ACK sip:*71203 at 192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.182:5060;branch=z9hG4bKc54f3d70528af1acd.fd80ab152c0418d30
Max-Forwards: 70
From: "T1201 Receptie" <sip:1201 at 192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203 at 192.168.1.18:5060>;tag=as32c804c2
Call-ID: dea0949d25f17307
CSeq: 28836 ACK
User-Agent: Aastra 53i/2.4.1.37
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
elspbx*CLI>
<--- SIP read from UDP://192.168.1.182:5060 --->
INVITE sip:*71203 at 192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.182:5060;branch=z9hG4bK3aa762215bd6dd755.562e74fab0af22fec
Max-Forwards: 70
From: "T1201 Receptie" <sip:1201 at 192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203 at 192.168.1.18:5060>
Call-ID: dea0949d25f17307
CSeq: 28837 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest
username="1201",realm="asterisk",nonce="38034d5d",uri="sip:*71203 at 192.168.1.18:5060",response="59e04b0c97616a5bfa4489e867c4c6b3",algorithm=MD5
Contact: "T1201 Receptie"
<sip:1201 at 192.168.1.182:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1A4BFE>"
Supported: gruu, timer, 100rel, replaces
User-Agent: Aastra 53i/2.4.1.37
Content-Type: application/sdp
Content-Length: 285
v=0
o=MxSIP 0 0 IN IP4 192.168.1.182
s=SIP Call
c=IN IP4 192.168.1.182
t=0 0
m=audio 3000 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 14 lines) ---
Sending to 192.168.1.182 : 5060 (no NAT)
Using INVITE request as basis request - dea0949d25f17307
Found peer '1201' for '1201' from 192.168.1.182:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.182:3000
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c
(ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.182:3000
Looking for *71203 in els-receptie (domain 192.168.1.18)
list_route: hop: <sip:1201 at 192.168.1.182:5060;transport=udp>
elspbx*CLI>
<--- Transmitting (no NAT) to 192.168.1.182:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.182:5060;branch=z9hG4bK3aa762215bd6dd755.562e74fab0af22fec;received=192.168.1.182
From: "T1201 Receptie" <sip:1201 at 192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203 at 192.168.1.18:5060>
Call-ID: dea0949d25f17307
CSeq: 28837 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: <sip:*71203 at 192.168.1.18>
Content-Length: 0
<------------>
-- Executing [*71203 at els-receptie:1] Set("SIP/1201-1c4be5c8", "1203")
in new stack
-- Executing [*71203 at els-receptie:2] ExtenSpy("SIP/1201-1c4be5c8",
"*71203,w") in new stack
Audio is at 192.168.1.18 port 10522
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
elspbx*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.182:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.182:5060;branch=z9hG4bK3aa762215bd6dd755.562e74fab0af22fec;received=192.168.1.182
From: "T1201 Receptie" <sip:1201 at 192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203 at 192.168.1.18:5060>;tag=as39909610
Call-ID: dea0949d25f17307
CSeq: 28837 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: <sip:*71203 at 192.168.1.18>
Content-Type: application/sdp
Content-Length: 330
v=0
o=root 12031972 12031972 IN IP4 192.168.1.18
s=Asterisk PBX 1.6.1.6
c=IN IP4 192.168.1.18
t=0 0
m=audio 10522 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
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
<------------>
elspbx*CLI>
<--- SIP read from UDP://192.168.1.182:5060 --->
ACK sip:*71203 at 192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.182:5060;branch=z9hG4bK059acb2e9930675bf.f1658ce4d7371f79f
Max-Forwards: 70
From: "T1201 Receptie" <sip:1201 at 192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203 at 192.168.1.18:5060>;tag=as39909610
Call-ID: dea0949d25f17307
CSeq: 28837 ACK
Authorization: Digest
username="1201",realm="asterisk",nonce="38034d5d",uri="sip:*71203 at 192.168.1.18:5060",response="59e04b0c97616a5bfa4489e867c4c6b3",algorithm=MD5
Supported: gruu
User-Agent: Aastra 53i/2.4.1.37
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
-- <SIP/1201-1c4be5c8> Playing 'beep.gsm' (language 'en')
-- <SIP/1201-1c4be5c8> Playing 'spy-sip.gsm' (language 'en')
-- <SIP/1201-1c4be5c8> Playing 'digits/1.gsm' (language 'en')
-- <SIP/1201-1c4be5c8> Playing 'digits/2.gsm' (language 'en')
-- <SIP/1201-1c4be5c8> Playing 'digits/0.gsm' (language 'en')
-- <SIP/1201-1c4be5c8> Playing 'digits/3.gsm' (language 'en')
== Spying on channel SIP/1203-018e21f8
[Oct 21 16:29:49] NOTICE[16197]: app_chanspy.c:292 start_spying: Attaching
SIP/1201-1c4be5c8 to SIP/1203-018e21f8
[Oct 21 16:29:49] NOTICE[16197]: app_chanspy.c:292 start_spying: Attaching
SIP/1201-1c4be5c8 to SIP/1203-018e21f8
[Oct 21 16:29:49] NOTICE[16197]: app_chanspy.c:292 start_spying: Attaching
SIP/1201-1c4be5c8 to SIP/1151-01a06b58
elspbx*CLI> sip set debug off
SIP Debugging Disabled
== Done Spying on channel SIP/1203-018e21f8
== Spawn extension (els-receptie, *71203, 2) exited non-zero on
'SIP/1201-1c4be5c8'
Issue History
Date Modified Username Field Change
======================================================================
2009-10-21 09:35 elsto Note Added: 0112523
======================================================================
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