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Summary:ASTERISK-15310: [regression] DTMF Tones not working
Reporter:Hendrik van der Ploeg (elsto)Labels:
Date Opened:2009-12-14 02:00:42.000-0600Date Closed:2011-06-07 14:01:04
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 1.6.1.9-dtmf-working.log
( 1) dtmf.log
( 2) Out-of-Band_DTMF_disabled.log
( 3) relevant-config.txt
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
Comments:By: Hendrik van der Ploeg (elsto) 2009-12-14 08:23:49.000-0600

<--- SIP read from UDP://192.168.1.65:5060 --->
INVITE sip:009000909@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@192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909@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@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@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@192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909@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@192.168.3.18>
Content-Length: 0


<------------>
set_destination: Parsing <sip:1902@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@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@192.168.3.18:5060>;tag=as60763b78
To: "T1902 Receptie2" <sip:1902@192.168.3.18:5060>;tag=d5ccb4f001
Contact: <sip:1902@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@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@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@192.168.3.18:5060>;tag=as60763b78
To: "T1902 Receptie2" <sip:1902@192.168.3.18:5060>;tag=d5ccb4f001
Call-ID: c16226de5daa5db0
CSeq: 106 NOTIFY
Contact: "T1902 Receptie2" <sip:1902@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@192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909@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@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@192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909@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@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@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@192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909@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@192.168.3.18:5060>;tag=9201fb26ae
To: <sip:1902@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@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@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@192.168.3.18:5060>;tag=9201fb26ae
To: <sip:1902@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@192.168.3.18:5060>;tag=9201fb26ae
To: <sip:1902@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@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@192.168.3.18:5060>;tag=9201fb26ae
To: <sip:1902@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@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@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@192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909@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@192.168.3.18:5060>;tag=14f9e8c911
To: "009000909" <sip:009000909@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@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@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@192.168.3.18:5060>;tag=as60763b78
To: "T1902 Receptie2" <sip:1902@192.168.3.18:5060>;tag=d5ccb4f001
Contact: <sip:1902@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@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@192.168.3.18:5060>;tag=as60763b78
To: "T1902 Receptie2" <sip:1902@192.168.3.18:5060>;tag=d5ccb4f001
Call-ID: c16226de5daa5db0
CSeq: 107 NOTIFY
Contact: "T1902 Receptie2" <sip:1902@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
By: rsw686 (rsw686) 2009-12-14 08:25:29.000-0600

Have a look at my issue relating to DTMF. If 1.6.1.7-rc1 works for you then it is the same problem. You could try applying my patch asterisk-1.6.1.11-constantssrc.patch and adding constantssrc=true to sip.conf.

https://issues.asterisk.org/view.php?id=16292
By: Hendrik van der Ploeg (elsto) 2009-12-14 08:36:50.000-0600

No luck, sorry
By: Leif Madsen (lmadsen) 2009-12-14 09:31:28.000-0600

Next time, please attach debug logs as a file to the issue, and not inline as it makes working with the issue much harder that way.
By: Hendrik van der Ploeg (elsto) 2009-12-14 09:33:13.000-0600

Yes, I noticed this option this afternoon for the first time. Sorry
By: Leif Madsen (lmadsen) 2009-12-14 09:34:09.000-0600

Additionally, I don't see any DTMF logging in your attachment. Have you enabled dtmf logging in logger.conf?
By: Hendrik van der Ploeg (elsto) 2009-12-14 09:43:20.000-0600

I've uploaded the log now with dtmf logging
By: Leif Madsen (lmadsen) 2009-12-14 12:31:22.000-0600

Well it looks like Asterisk is receiving the DTMF and passing it through. Can you also attach the relevant parts of your sip.conf file, and which model of phone you're using? That may be needed for a developer to reproduce the issue.

I'm not sure what else beyond that to ask for in this instance.
By: Leif Madsen (lmadsen) 2009-12-14 12:33:13.000-0600

Oh, I just thought of something. What version of Asterisk were you running previously that didn't have this issue? Did you change anything in your configuration?

If using an earlier version of 1.6.1.11 worked, could you step back versions until you find the latest possible version that it worked in, and which version first caused the regression?

It'd be nice to try and track down where this possible regression came in. I'd like to know for sure that the EXACT same setup and scenario that you're reporting this against works with another version of Asterisk.
By: Hendrik van der Ploeg (elsto) 2009-12-15 02:38:50.000-0600

Ok I just found out that;

with version 1.6.1.10 a voicetape (or whatever you call it) isn't hearable.
So I even wasn't able to test the DTMF tones because I couldn't hear the voicetape.

With version 1.6.1.9 I can hear a voicetape and I can also send DTMF tone.
I've uploaded a log of version 1.6.1.9

Changes I made:

Version 1.6.1.10 I added 'parkinglot' in 'sip.conf'

Version 1.6.1.11 I added 't38pt_udptl = yes,fec,maxdatagram=400' in 'sip.conf'

I use Aastra SIP phones.
The 53i and 31i


By: Leif Madsen (lmadsen) 2009-12-15 07:22:41.000-0600

OK, what is your topology? What are you connecting to, what devices, etc... ? There isn't a whole lot of information to go on here, so the more information about what has changed, what your topology is, etc... will certainly help.

Additionally, you may want to attach the relevant parts of your sip.conf and extensions.conf files here in order to reproduce the issue. Knowing how you're connecting (via which provider, etc..) will be paramount to tracking down this scenario.
By: Hendrik van der Ploeg (elsto) 2009-12-15 07:53:28.000-0600

It's a normal star-topology. (I assume you mean network-layout)
I'm calling out via a SIP-trunk of a Dutch voip provider.

Checkout the uploaded file for -what I think is- the relevant config

Aastra ==>  Asterisk(IBM x-server) ===>  CiscoRouter (which belongs to provider)
By: Hendrik van der Ploeg (elsto) 2009-12-18 04:00:11.000-0600

Additionally I've added the log file with the setting;

"Force RFC2833 Out-of-Band DTMF" turned off on my Aastra SIP phone.

When this option is disabled the DTMF tones work.
By: Terry Wilson (twilson) 2010-03-12 19:01:40.000-0600

Could you go ahead and update to the latest 1.6.1 from SVN and test to see if this is fixed now? I have made a change that I think might affect this bug.
By: Leif Madsen (lmadsen) 2010-03-15 13:50:20

Set to feedback while we await word from the reporter.
By: Paul Belanger (pabelanger) 2010-04-28 15:59:59

Ping! Wanted to see if this was still an issue after you tested with the latest 1.6.1 branch.
By: Paul Belanger (pabelanger) 2010-05-12 13:18:29

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.

Further information can be found at http://www.asterisk.org/developers/bug-guidelines