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Summary:ASTERISK-15019: Application Extenspy
Reporter:Hendrik van der Ploeg (elsto)Labels:
Date Opened:2009-10-21 07:15:42Date Closed:2009-11-09 15:07:39.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Applications/app_chanspy
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
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
Comments:By: Leif Madsen (lmadsen) 2009-10-21 09:29:37

More information is required to move this issue forward. Please provide:

* Console output
* Dialplan configuration
* ...anything else relevant to reproducing this issue
By: Hendrik van der Ploeg (elsto) 2009-10-21 09:35:53

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@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@192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203@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@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@192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203@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@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@192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203@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@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@192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203@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@192.168.1.18:5060",response="59e04b0c97616a5bfa4489e867c4c6b3",algorithm=MD5
Contact: "T1201 Receptie" <sip:1201@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@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@192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203@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@192.168.1.18>
Content-Length: 0


<------------>
   -- Executing [*71203@els-receptie:1] Set("SIP/1201-1c4be5c8", "1203") in new stack
   -- Executing [*71203@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@192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203@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@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@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@192.168.1.18:5060>;tag=2d34363831
To: "*71203" <sip:*71203@192.168.1.18:5060>;tag=as39909610
Call-ID: dea0949d25f17307
CSeq: 28837 ACK
Authorization: Digest username="1201",realm="asterisk",nonce="38034d5d",uri="sip:*71203@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'
By: Leif Madsen (lmadsen) 2009-11-09 15:07:39.000-0600

I'm going to call this a duplicate of ASTERISK-15077 and close this issue. Please continue to monitor this issue there.