[asterisk-bugs] [Asterisk 0017169]: no sound on Playback(<file>, noanswer)
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Apr 12 17:12:34 CDT 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=17169
======================================================================
Reported By: leroi05
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 17169
Category: Applications/app_playback
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.6.2.7-rc1
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-04-12 12:24 CDT
Last Modified: 2010-04-12 17:12 CDT
======================================================================
Summary: no sound on Playback(<file>,noanswer)
Description:
There is no sound if I try to playback a file without answering the channel
(sip, iax). Tried versoins 1.6.2.5, 1.6.2.6, 1.6.2.7-rc1, 1.6.1.19-rc1,
1.6.0.27-rc1.
On sip channel rtp debug gives nothing.
Reproduceable with minimal dialplan:
exten => _X.,1,Playback(demo-congrats,noanswer)
======================================================================
----------------------------------------------------------------------
(0120305) leroi05 (reporter) - 2010-04-12 17:12
https://issues.asterisk.org/view.php?id=17169#c120305
----------------------------------------------------------------------
Well, I thought it's pretty clear what I did and what I've got.. but ok..
1.
-- download asterisk tarball of any version I've pointed
-- unpack/configure/make/make install/start
-- add sip friend in /etc/asterisk/sip.conf which looks like this:
[test]
secret=test
host=dynamic
type=friend
context=noanswer
-- register sip client with this account (see if it registered with sip
show peers)
-- add context in /etc/asterisk/extensions.conf:
[noanswer]
exten => _X.,1,Playback(demo-congrats,noanswer)
-- dial something like 55
-- listen to the silence
2. I expected to hear demo-congrats file (this works on our production
server with 1.4.21.2 as I also stated)
3. I hear nothing (no sound)
Console output:
==========================================================================
*CLI> sip set debug peer test
SIP Debugging Enabled for IP: 192.168.1.106:2049
orsus*CLI> rtp set debug on
RTP Debugging Enabled
*CLI>
<--- SIP read from UDP://192.168.1.106:2049 --->
INVITE sip:99 at 192.168.1.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:2049;branch=z9hG4bK-jks72ma6zncu;rport
From: "test" <sip:test at 192.168.1.3>;tag=i43s1vktnt
To: <sip:99 at 192.168.1.3;user=phone>
Call-ID: 3c26fca9c35c-jhhgkvo7xql8
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:test at 192.168.1.106:2049;line=3suaez42>;reg-id=1
X-Serialnumber: 000413291821
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/8.2.25
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 473
v=0
o=root 259342033 259342033 IN IP4 192.168.1.106
s=call
c=IN IP4 192.168.1.106
t=0 0
m=audio 62414 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:r1+W3cQjCCCZdiCXML6CH/fZ8YbqOuqHRbom/t1/
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=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (19 headers 19 lines) ---
== Using SIP RTP CoS mark 5
Sending to 192.168.1.106 : 2049 (no NAT)
Using INVITE request as basis request - 3c26fca9c35c-jhhgkvo7xql8
Found user 'test' for 'test'
*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.106:2049 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.106:2049;branch=z9hG4bK-jks72ma6zncu;received=192.168.1.106;rport=2049
From: "test" <sip:test at 192.168.1.3>;tag=i43s1vktnt
To: <sip:99 at 192.168.1.3;user=phone>;tag=as5ab0dc35
Call-ID: 3c26fca9c35c-jhhgkvo7xql8
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0.27-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="20e481c0"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '3c26fca9c35c-jhhgkvo7xql8' in 32000
ms (Method: INVITE)
*CLI>
<--- SIP read from UDP://192.168.1.106:2049 --->
ACK sip:99 at 192.168.1.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:2049;branch=z9hG4bK-jks72ma6zncu;rport
From: "test" <sip:test at 192.168.1.3>;tag=i43s1vktnt
To: <sip:99 at 192.168.1.3;user=phone>;tag=as5ab0dc35
Call-ID: 3c26fca9c35c-jhhgkvo7xql8
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:test at 192.168.1.106:2049;line=3suaez42>;reg-id=1
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
*CLI>
<--- SIP read from UDP://192.168.1.106:2049 --->
INVITE sip:99 at 192.168.1.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:2049;branch=z9hG4bK-f97pfdqqcm9w;rport
From: "test" <sip:test at 192.168.1.3>;tag=i43s1vktnt
To: <sip:99 at 192.168.1.3;user=phone>
Call-ID: 3c26fca9c35c-jhhgkvo7xql8
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:test at 192.168.1.106:2049;line=3suaez42>;reg-id=1
X-Serialnumber: 000413291821
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/8.2.25
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest
username="test",realm="asterisk",nonce="20e481c0",uri="sip:99 at 192.168.1.3;user=phone",response="c1eb36c3a8d275eec431e8ceb051bf77",algorithm=MD5
Content-Type: application/sdp
Content-Length: 473
v=0
o=root 259342033 259342033 IN IP4 192.168.1.106
s=call
c=IN IP4 192.168.1.106
t=0 0
m=audio 62414 RTP/AVP 0 8 9 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:r1+W3cQjCCCZdiCXML6CH/fZ8YbqOuqHRbom/t1/
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=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (20 headers 19 lines) ---
Sending to 192.168.1.106 : 2049 (no NAT)
Using INVITE request as basis request - 3c26fca9c35c-jhhgkvo7xql8
Found user 'test' for 'test'
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 - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x190f
(g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing),
combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.106:62414
Looking for 99 in noanswer (domain 192.168.1.3)
list_route: hop: <sip:test at 192.168.1.106:2049;line=3suaez42>
*CLI>
<--- Transmitting (no NAT) to 192.168.1.106:2049 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.106:2049;branch=z9hG4bK-f97pfdqqcm9w;received=192.168.1.106;rport=2049
From: "test" <sip:test at 192.168.1.3>;tag=i43s1vktnt
To: <sip:99 at 192.168.1.3;user=phone>
Call-ID: 3c26fca9c35c-jhhgkvo7xql8
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.27-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
ontact: <sip:99 at 192.168.1.3>
Content-Length: 0
<------------>
-- Executing [99 at noanswer:1] Playback("SIP/test-00000002",
"demo-congrats,noanswer") in new stack
-- <SIP/test-00000002> Playing 'demo-congrats.alaw' (language 'en')
*CLI>
==========================================================================
Here I hangup because there is no point in waiting:
==========================================================================
<--- SIP read from UDP://192.168.1.106:2049 --->
CANCEL sip:99 at 192.168.1.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:2049;branch=z9hG4bK-f97pfdqqcm9w;rport
From: "test" <sip:test at 192.168.1.3>;tag=i43s1vktnt
To: <sip:99 at 192.168.1.3;user=phone>
Call-ID: 3c26fca9c35c-jhhgkvo7xql8
CSeq: 2 CANCEL
Max-Forwards: 70
Reason: SIP;cause=487;text="Request terminated by user"
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.1.106 : 2049 (no NAT)
*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.106:2049 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
192.168.1.106:2049;branch=z9hG4bK-f97pfdqqcm9w;received=192.168.1.106;rport=2049
From: "test" <sip:test at 192.168.1.3>;tag=i43s1vktnt
To: <sip:99 at 192.168.1.3;user=phone>;tag=as4bc69dc6
Call-ID: 3c26fca9c35c-jhhgkvo7xql8
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.27-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
*CLI>
<--- Transmitting (no NAT) to 192.168.1.106:2049 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.106:2049;branch=z9hG4bK-f97pfdqqcm9w;received=192.168.1.106;rport=2049
From: "test" <sip:test at 192.168.1.3>;tag=i43s1vktnt
To: <sip:99 at 192.168.1.3;user=phone>;tag=as4bc69dc6
Call-ID: 3c26fca9c35c-jhhgkvo7xql8
CSeq: 2 CANCEL
User-Agent: Asterisk PBX 1.6.0.27-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (noanswer, 99, 1) exited non-zero on
'SIP/test-00000002'
*CLI>
<--- SIP read from UDP://192.168.1.106:2049 --->
ACK sip:99 at 192.168.1.3;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:2049;branch=z9hG4bK-f97pfdqqcm9w;rport
From: "test" <sip:test at 192.168.1.3>;tag=i43s1vktnt
To: <sip:99 at 192.168.1.3;user=phone>;tag=as4bc69dc6
Call-ID: 3c26fca9c35c-jhhgkvo7xql8
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:test at 192.168.1.106:2049;line=3suaez42>;reg-id=1
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '3c26fca9c35c-jhhgkvo7xql8' Method: ACK
===========================================================================
It's very clear and easy to reproduce - I don't understand your "Lack of
information".
Issue History
Date Modified Username Field Change
======================================================================
2010-04-12 17:12 leroi05 Note Added: 0120305
======================================================================
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