[asterisk-bugs] [Asterisk 0016438]: DTMF Tones not working
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Dec 14 08:23:49 CST 2009
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=16438
======================================================================
Reported By: elsto
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 16438
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.6.1.11
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-12-14 02:00 CST
Last Modified: 2009-12-14 08:23 CST
======================================================================
Summary: DTMF Tones not working
Description:
Hello,
With Asterisk version 1.6.1.11 the sending of DTMF rfc2833 tones isn't
working anymore.
But when I disable the option "Force RFC2833 Out-of-Band DTMF" on my SIP
phone it works fine.
Is this a planned change or simply a bug?
Regards,
Hendrik
======================================================================
----------------------------------------------------------------------
(0115190) elsto (reporter) - 2009-12-14 08:23
https://issues.asterisk.org/view.php?id=16438#c115190
----------------------------------------------------------------------
<--- SIP read from UDP://192.168.1.65:5060 --->
INVITE sip:009000909 at 192.168.3.18:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bKe60b193044cc34378.28be0b5dd310ef713
Max-Forwards: 70
From: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909 at 192.168.3.18:5060>
Call-ID: 63c575f2bb897fe3
CSeq: 31331 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "T1902 Receptie2"
<sip:1902 at 192.168.1.65:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D22C1CE>"
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 6731i/2.5.2.1010
Content-Type: application/sdp
Content-Length: 283
v=0
o=MxSIP 0 0 IN IP4 192.168.1.65
s=SIP Call
c=IN IP4 192.168.1.65
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 192.168.1.65 : 5060 (no NAT)
Using INVITE request as basis request - 63c575f2bb897fe3
Found peer '1902' for '1902' from 192.168.1.65:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
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 192.168.1.65:3000
Looking for 009000909 in asn-receptie (domain 192.168.3.18)
list_route: hop: <sip:1902 at 192.168.1.65:5060;transport=udp>
<--- Transmitting (no NAT) to 192.168.1.65:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bKe60b193044cc34378.28be0b5dd310ef713;received=192.168.1.65
From: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909 at 192.168.3.18:5060>
Call-ID: 63c575f2bb897fe3
CSeq: 31331 INVITE
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:009000909 at 192.168.3.18>
Content-Length: 0
<------------>
set_destination: Parsing <sip:1902 at 192.168.1.65:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.1.65, port 5060
Reliably Transmitting (no NAT) to 192.168.1.65:5060:
NOTIFY sip:1902 at 192.168.1.65:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.3.18:5060;branch=z9hG4bK42f17c9e;rport
Max-Forwards: 70
From: "" <sip:1902 at 192.168.3.18:5060>;tag=as60763b78
To: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=d5ccb4f001
Contact: <sip:1902 at 192.168.3.18>
Call-ID: c16226de5daa5db0
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX 1.6.1.11
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 209
<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="4"
state="full" entity="sip:1902 at 192.168.3.18:5060">
<dialog id="1902">
<state>confirmed</state>
</dialog>
</dialog-info>
---
== Extension Changed 1902[blf] new state Busy for Notify User 1902
-- Executing [009000909 at asn-receptie:1] Dial("SIP/1902-0000002c",
"SIP/Priority_out/09000909,,T") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called Priority_out/09000909
wglpbx*CLI>
<--- SIP read from UDP://192.168.1.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.3.18:5060;branch=z9hG4bK42f17c9e;rport=5060;received=192.168.3.18
From: "" <sip:1902 at 192.168.3.18:5060>;tag=as60763b78
To: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=d5ccb4f001
Call-ID: c16226de5daa5db0
CSeq: 106 NOTIFY
Contact: "T1902 Receptie2"
<sip:1902 at 192.168.1.65:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D22C1CE>"
Server: Aastra 6731i/2.5.2.1010
Content-Length: 0
wglpbx*CLI>
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
-- SIP/Priority_out-0000002d is making progress passing it to
SIP/1902-0000002c
Audio is at 192.168.3.18 port 16690
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
<--- Transmitting (no NAT) to 192.168.1.65:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bKe60b193044cc34378.28be0b5dd310ef713;received=192.168.1.65
From: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909 at 192.168.3.18:5060>;tag=as1d612409
all-ID: 63c575f2bb897fe3
CSeq: 31331 INVITE
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:009000909 at 192.168.3.18>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 101265788 101265788 IN IP4 192.168.3.18
s=Asterisk PBX 1.6.1.11
c=IN IP4 192.168.3.18
t=0 0
m=audio 16690 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
<------------>
-- SIP/Priority_out-0000002d is making progress passing it to
SIP/1902-0000002c
-- SIP/Priority_out-0000002d answered SIP/1902-0000002c
Audio is at 192.168.3.18 port 16690
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
<--- Reliably Transmitting (no NAT) to 192.168.1.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bKe60b193044cc34378.28be0b5dd310ef713;received=192.168.1.65
From: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909 at 192.168.3.18:5060>;tag=as1d612409
Call-ID: 63c575f2bb897fe3
CSeq: 31331 INVITE
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:009000909 at 192.168.3.18>
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 101265788 101265789 IN IP4 192.168.3.18
s=Asterisk PBX 1.6.1.11
c=IN IP4 192.168.3.18
t=0 0
m=audio 16690 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
<------------>
wglpbx*CLI>
<--- SIP read from UDP://192.168.1.65:5060 --->
ACK sip:009000909 at 192.168.3.18 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bK2e9061dbcf7936455.10f227a4ec364db3f
Max-Forwards: 70
From: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909 at 192.168.3.18:5060>;tag=as1d612409
Call-ID: 63c575f2bb897fe3
CSeq: 31331 ACK
User-Agent: Aastra 6731i/2.5.2.1010
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
wglpbx*CLI>
<--- SIP read from UDP://192.168.1.65:5060 --->
REGISTER sip:192.168.3.18:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bKe29aa9249e40f1c54.532d467496c8505bb
Max-Forwards: 70
From: <sip:1902 at 192.168.3.18:5060>;tag=9201fb26ae
To: <sip:1902 at 192.168.3.18:5060>
Call-ID: 0d5e8ab61a60931c
CSeq: 26871 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest
username="1902",realm="asterisk",nonce="60421311",uri="sip:192.168.3.18:5060",response="8364e10c6bd8617d5fd4ac7447f0f4c4",algorithm=MD5
Contact: "T1902 Receptie2"
<sip:1902 at 192.168.1.65:5060;transport=udp>;expires=30;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D22C1CE>"
Supported: gruu, path
User-Agent: Aastra 6731i/2.5.2.1010
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.1.65 : 5060 (no NAT)
[Dec 14 15:21:59] NOTICE[22891]: chan_sip.c:11847 check_auth: Correct
auth, but based on stale nonce received from
'<sip:1902 at 192.168.3.18:5060>'
<--- Transmitting (no NAT) to 192.168.1.65:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bKe29aa9249e40f1c54.532d467496c8505bb;received=192.168.1.65
From: <sip:1902 at 192.168.3.18:5060>;tag=9201fb26ae
To: <sip:1902 at 192.168.3.18:5060>;tag=as54a8f456
Call-ID: 0d5e8ab61a60931c
CSeq: 26871 REGISTER
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="1a9469c8", stale=true
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '0d5e8ab61a60931c' in 32000 ms
(Method: REGISTER)
wglpbx*CLI>
<--- SIP read from UDP://192.168.1.65:5060 --->
REGISTER sip:192.168.3.18:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bKf939bfc919088d625.1ce8b5bd9f8196719
Max-Forwards: 70
From: <sip:1902 at 192.168.3.18:5060>;tag=9201fb26ae
To: <sip:1902 at 192.168.3.18:5060>
Call-ID: 0d5e8ab61a60931c
CSeq: 26872 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest
username="1902",realm="asterisk",nonce="1a9469c8",uri="sip:192.168.3.18:5060",response="6bc9fbadb00af6fe5f2bf28be276de92",algorithm=MD5
Contact: "T1902 Receptie2"
<sip:1902 at 192.168.1.65:5060;transport=udp>;expires=30;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D22C1CE>"
Supported: gruu, path
User-Agent: Aastra 6731i/2.5.2.1010
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.1.65 : 5060 (no NAT)
wglpbx*CLI>
<--- Transmitting (no NAT) to 192.168.1.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bKf939bfc919088d625.1ce8b5bd9f8196719;received=192.168.1.65
From: <sip:1902 at 192.168.3.18:5060>;tag=9201fb26ae
To: <sip:1902 at 192.168.3.18:5060>;tag=as54a8f456
Call-ID: 0d5e8ab61a60931c
CSeq: 26872 REGISTER
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 30
Contact: sip:1902 at 192.168.1.65:5060;transport=udp;expires=30
Date: Mon, 14 Dec 2009 14:21:59 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '0d5e8ab61a60931c' in 32000 ms
(Method: REGISTER)
wglpbx*CLI>
<--- SIP read from UDP://192.168.1.65:5060 --->
BYE sip:009000909 at 192.168.3.18 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bK080bcbcf57e733e91.752f22336186d5c8b
Max-Forwards: 70
From: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909 at 192.168.3.18:5060>;tag=as1d612409
Call-ID: 63c575f2bb897fe3
CSeq: 31332 BYE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Supported: gruu, path, timer
User-Agent: Aastra 6731i/2.5.2.1010
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.1.65 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.1.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.65:5060;branch=z9hG4bK080bcbcf57e733e91.752f22336186d5c8b;received=192.168.1.65
From: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909 at 192.168.3.18:5060>;tag=as1d612409
Call-ID: 63c575f2bb897fe3
CSeq: 31332 BYE
Server: Asterisk PBX 1.6.1.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (asn-receptie, 009000909, 1) exited non-zero on
'SIP/1902-0000002c'
set_destination: Parsing <sip:1902 at 192.168.1.65:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.1.65, port 5060
Reliably Transmitting (no NAT) to 192.168.1.65:5060:
NOTIFY sip:1902 at 192.168.1.65:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.3.18:5060;branch=z9hG4bK55bd0292;rport
Max-Forwards: 70
From: "" <sip:1902 at 192.168.3.18:5060>;tag=as60763b78
To: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=d5ccb4f001
Contact: <sip:1902 at 192.168.3.18>
Call-ID: c16226de5daa5db0
CSeq: 107 NOTIFY
User-Agent: Asterisk PBX 1.6.1.11
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 210
<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="5"
state="full" entity="sip:1902 at 192.168.3.18:5060">
<dialog id="1902">
<state>terminated</state>
</dialog>
</dialog-info>
---
== Extension Changed 1902[blf] new state Idle for Notify User 1902
Really destroying SIP dialog '63c575f2bb897fe3' Method: BYE
wglpbx*CLI>
<--- SIP read from UDP://192.168.1.65:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.3.18:5060;branch=z9hG4bK55bd0292;rport=5060;received=192.168.3.18
From: "" <sip:1902 at 192.168.3.18:5060>;tag=as60763b78
To: "T1902 Receptie2" <sip:1902 at 192.168.3.18:5060>;tag=d5ccb4f001
Call-ID: c16226de5daa5db0
CSeq: 107 NOTIFY
Contact: "T1902 Receptie2"
<sip:1902 at 192.168.1.65:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D22C1CE>"
Server: Aastra 6731i/2.5.2.1010
Content-Length: 0
wglpbx*CLI>
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
wglpbx*CLI> sip set debug off
SIP Debugging Disabled
Issue History
Date Modified Username Field Change
======================================================================
2009-12-14 08:23 elsto Note Added: 0115190
======================================================================
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