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Summary:ASTERISK-00924: DTMF not being passed when IVR does not send "Connect" status
Reporter:sdolloff (sdolloff)Labels:
Date Opened:2004-01-27 11:41:08.000-0600Date Closed:2008-01-15 14:45:23.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) debugtoasterisk.txt
( 1) debugtocisco.txt
( 2) sipuratoasterisk.txt
( 3) sipuratocisco.txt
Description:DTMF does not get passed via INFO, rfc2833, or inband when certain IVR systems pick up the call.  There are multiple bugs related to this item that have not been truly resolved including ASTERISK-1600204.  I am using an AS5350 Cisco gateway and SPA-2000 ATAs.  I have also tried using ATA-186.  If I connect the SPA to the AS5350 directly, it works fine using any of the dtmf relay modes.

****** ADDITIONAL INFORMATION ******

One specific example of this type of IVR is 1-800-882-8880

Response regarding this from Cisco:

SIP phones not able to break the audio from certain IVR providers. The problem is that until we get the PRI Connect message, the gateway will not send the SIP 200 OK.
There is no workaround. We are at the mercy of the far end on these calls. They need to actually answer the call (True Answer) so that the network can send back the Connect message.  They recommended using INFO or inband, either works fine until you put * in the middle.

Response regarding this from Sipura:

INFO can only be sent when 200 OK is received; that is, when the dialog
has been confirmed.

But the SPA-2000 starts sending inband audio (RTP) after receiving 183 with SDP (not 180)
and you should be able to easily confirm that by capturing the RTP packets during ringing. In fact,
SPA-2000 can sent AVT tone (instead of inband) DTMF via RTP during this time.

Many IVR system (such as Delta Airlines) are activated right after the 183 (but before 200) and
the SPA-2000 works just fine with, for example, Voicepulse or Vonage services.

Capture of sip debug during call:

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 209.242.15.34:5060;branch=z9hG4bK0eb747ff
From: "8477190222" <sip:8477190222@209.242.15.34>;tag=as0a3389cc
To: <sip:18008828880@209.242.19.118>;tag=1E1D3C80-6E4
Date: Fri, 12 May 2000 20:58:45 GMT
Call-ID: 67ffcc91416477f6329837f76be74664@209.242.15.34
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 216

v=0
o=CiscoSystemsSIP-GW-UserAgent 6411 8850 IN IP4 209.242.19.118
s=SIP Call
c=IN IP4 209.242.19.118
t=0 0
m=audio 18810 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

12 headers, 9 lines
   -- SIP/VGW01-a0cd is making progress passing it to SIP/8477190222-04a3
We're at 209.242.15.34 port 18550
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
Transmitting (NAT):
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 216.145.234.122:5060;branch=z9hG4bK-e6f18d9c;received=216.145.234.122
From: <sip:8477190222@voip2.dls.net>;tag=8fbd507be1da3553
To: <sip:18008828880@voip2.dls.net>;tag=as35b51bd2
Call-ID: db907c84-6f9a727f@216.145.234.122
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:18008828880@209.242.15.34>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 22665 22665 IN IP4 209.242.15.34
s=session
c=IN IP4 209.242.15.34
t=0 0
m=audio 18550 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

to 216.145.234.122:5060
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:209.242.5.25 SIP/2.0
Via: SIP/2.0/UDP 209.242.15.34:5060;branch=z9hG4bK60767cc9
From: "asterisk" <sip:asterisk@209.242.15.34>;tag=as38ee8fce
To: <sip:209.242.5.25>
Contact: <sip:asterisk@209.242.15.34>
Call-ID: 4182cb0b140ffde8339bfc0129f8caa6@209.242.15.34
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 27 Jan 2004 17:23:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

(no NAT) to 209.242.5.25:5060
voip2*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.242.15.34:5060;branch=z9hG4bK60767cc9
From: "asterisk" <sip:asterisk@209.242.15.34>;tag=as38ee8fce
To: <sip:209.242.5.25>;tag=673209317
Call-ID: 4182cb0b140ffde8339bfc0129f8caa6@209.242.15.34
CSeq: 102 OPTIONS
Server: Cisco ATA 186  v2.16.2.TEST ata18x (031117a)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 251
Content-Type: application/sdp

v=0
o=8476499564 46126 46126 IN IP4 10.1.18.28
s=ATA186 Call
c=IN IP4 209.242.5.25
t=0 0
m=audio 16384 RTP/AVP 0 8 4 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

10 headers, 11 lines
voip2*CLI>

Sip read:
CANCEL sip:18008828880@voip2.dls.net SIP/2.0
Via: SIP/2.0/UDP 216.145.234.122:5060;branch=z9hG4bK-30dad857
From: <sip:8477190222@voip2.dls.net>;tag=8fbd507be1da3553
To: <sip:18008828880@voip2.dls.net>
Call-ID: db907c84-6f9a727f@216.145.234.122
CSeq: 102 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest username="8477190222",realm="asterisk",nonce="4437665d",uri="sip:18008828880@voip2.dls.net",algorithm=MD5,response="8a617ead009a1a8bbbe2dbde4751891b"
User-Agent: Sipura/SPA2000-1.0.24
Content-Length: 0


10 headers, 0 lines
Sending to 216.145.234.122 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.145.234.122:5060;branch=z9hG4bK-30dad857;received=216.145.234.122
From: <sip:8477190222@voip2.dls.net>;tag=8fbd507be1da3553
To: <sip:18008828880@voip2.dls.net>;tag=as35b51bd2
Call-ID: db907c84-6f9a727f@216.145.234.122
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:18008828880@209.242.15.34>
Content-Length: 0


to 216.145.234.122:5060
Reliably Transmitting (NAT):
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 216.145.234.122:5060;branch=z9hG4bK-e6f18d9c;received=216.145.234.122
From: <sip:8477190222@voip2.dls.net>;tag=8fbd507be1da3553
To: <sip:18008828880@voip2.dls.net>;tag=as35b51bd2
Call-ID: db907c84-6f9a727f@216.145.234.122
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:18008828880@209.242.15.34>
Content-Length: 0


to 216.145.234.122:5060
Reliably Transmitting:
CANCEL sip:18008828880@209.242.19.118 SIP/2.0
Via: SIP/2.0/UDP 209.242.15.34:5060;branch=z9hG4bK0eb747ff
From: "8477190222" <sip:8477190222@209.242.15.34>;tag=as0a3389cc
To: <sip:18008828880@209.242.19.118>
Contact: <sip:8477190222@209.242.15.34>
Call-ID: 67ffcc91416477f6329837f76be74664@209.242.15.34
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 209.242.19.118:5060
 == Spawn extension (unlimited, 8008828880, 2) exited non-zero on 'SIP/8477190222-04a3'
voip2*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.242.15.34:5060;branch=z9hG4bK0eb747ff
From: "8477190222" <sip:8477190222@209.242.15.34>;tag=as0a3389cc
To: <sip:18008828880@209.242.19.118>
Date: Fri, 12 May 2000 20:58:56 GMT
Call-ID: 67ffcc91416477f6329837f76be74664@209.242.15.34
Content-Length: 0
CSeq: 102 CANCEL


8 headers, 0 lines
voip2*CLI>

Sip read:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 209.242.15.34:5060;branch=z9hG4bK0eb747ff
From: "8477190222" <sip:8477190222@209.242.15.34>;tag=as0a3389cc
To: <sip:18008828880@209.242.19.118>;tag=1E1D3C80-6E4
Date: Fri, 12 May 2000 20:58:56 GMT
Call-ID: 67ffcc91416477f6329837f76be74664@209.242.15.34
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


10 headers, 0 lines
Transmitting:
ACK sip:18008828880@209.242.19.118 SIP/2.0
Via: SIP/2.0/UDP 209.242.15.34:5060;branch=z9hG4bK0eb747ff
From: "8477190222" <sip:8477190222@209.242.15.34>;tag=as0a3389cc
To: <sip:18008828880@209.242.19.118>;tag=1E1D3C80-6E4
Contact: <sip:8477190222@209.242.15.34>
Call-ID: 67ffcc91416477f6329837f76be74664@209.242.15.34
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 209.242.19.118:5060
voip2*CLI>

Sip read:
ACK sip:18008828880@voip2.dls.net SIP/2.0
Via: SIP/2.0/UDP 216.145.234.122:5060;branch=z9hG4bK-e6f18d9c
From: <sip:8477190222@voip2.dls.net>;tag=8fbd507be1da3553
To: <sip:18008828880@voip2.dls.net>;tag=as35b51bd2
Call-ID: db907c84-6f9a727f@216.145.234.122
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="8477190222",realm="asterisk",nonce="4437665d",uri="sip:18008828880@voip2.dls.net",algorithm=MD5,response="e889bcb47645c7e334624d543a1b838b"
Contact: <sip:8477190222@216.145.234.122:5060>
User-Agent: Sipura/SPA2000-1.0.24
Content-Length: 0

I am running * 0.7.0 with the p1 patch.

Comments:By: Brian West (bkw918) 2004-02-02 00:40:06.000-0600

Happen to have any zaptel interfaces to test a zap to sip call?
By: sdolloff (sdolloff) 2004-02-02 11:06:42.000-0600

No.  This is our first deployment of Asterisk.  We do plan on deploying multiple smaller systems with Zaptel interfaces, but not until we can verify that the software is working correctly.

BKW, we talked on IRC and you briefly looked at the AS5350 configuration and then told me to talk to kram about it.  Kram told me that ticket 204 had already been resolved and I should open a ticket if I have a problem.

What further information can I provide to assist in troubleshooting this issue?
By: Brian West (bkw918) 2004-02-02 12:30:44.000-0600

try latest CVS... and then we will see.
By: Brian West (bkw918) 2004-02-02 12:38:25.000-0600

I need your conf files and such also.  Also what IOS version you have on the 5300 and have you checked cisco to see if it has any open issues?
By: sdolloff (sdolloff) 2004-02-03 09:35:37.000-0600

OK, updated from CVS as of 2/2/04

Currently running Version 12.2(13)T9.  Have also tried various versions of 12.3 which did not fix the problem, but did cause other ones.

I have an open ticket with Cisco.  The engineer is very interested in seeing this issue resolved, but both of us were able to solve this issue by removing * from the middle of the connection.

relevant cisco config:

dial-peer voice 1 pots
preference 1
destination-pattern .
no digit-strip
port 3/1:D
!
!
dial-peer voice 50 voip
incoming called-number .
no modem passthrough
session protocol sipv2
dtmf-relay rtp-nte
codec g711ulaw
fax rate disable
fax protocol pass-through g711ulaw
no vad
!

-----------sip.conf--------------
[general]
port = 5060                     ; Port to bind to
context = default               ; Default for incoming calls
dtmfmode=rfc2833

[VGW01]
type=friend
nat=no
host=209.242.1.1
context=default

[8477190222]
type=friend
secret=1234
nat=yes
host=dynamic
canreinvite=no
qualify=yes
mailbox=8477190222
context=unlimited

----------------extensions.conf--------------------

[unlimited]
include => on-net
include => international

[on-net]
exten => _NXXNXXXXXX,1,NoOp
exten => _1NXXNXXXXXX,1,StripMSD(1)

#include on_net.ext.2

[international]
include => longdistance
exten => _011.,1,Dial(Sip/${EXTEN}@VGW02)
exten => _011.,2,Congestion

[longdistance]

exten => _800NXXXXXX,2,Dial(SIP/1${EXTEN}@VGW01)
exten => _866NXXXXXX,2,Dial(SIP/1${EXTEN}@VGW01)
exten => _877NXXXXXX,2,Dial(SIP/1${EXTEN}@VGW01)
exten => _888NXXXXXX,2,Dial(SIP/1${EXTEN}@VGW01)
exten => _855NXXXXXX,2,Dial(SIP/1${EXTEN}@VGW01)
exten => _844NXXXXXX,2,Dial(SIP/1${EXTEN}@VGW01)
exten => _833NXXXXXX,2,Dial(SIP/1${EXTEN}@VGW01)
exten => _822NXXXXXX,2,Dial(SIP/1${EXTEN}@VGW01)

exten => _312NXXXXXX,2,Dial(SIP/1${EXTEN}@VGW01)
exten => _312NXXXXXX,3,Dial(SIP/1${EXTEN}@VGW02)

exten => _NXXNXXXXXX,2,Dial(SIP/1${EXTEN}@VGW02)

edited on: 02-03-04 09:36
By: Brian West (bkw918) 2004-02-03 18:59:38.000-0600

Can you get me a sipdebug when you make the call from asterisk then one when you make a call from the sipura and we will compair the two.  We will see where the message is missing and try to fix it.
By: sdolloff (sdolloff) 2004-02-04 12:04:51.000-0600

sipuratoasterisk.txt is an ethereal capture of udp 5060 during call through asterisk at ATA side.
sipuratocisco.txt is an ethereal capture of udp 5060 during call direct to AS5350 at ATA side
debugtoasterisk.txt are sipura debug and network captures during call through asterisk
debugtocisco.txt are sipura debug and network capture(only 1 way) during call direct to cisco

Please let me know what else I can provide
By: sdolloff (sdolloff) 2004-02-16 11:07:51.000-0600

Anyone had a chance to look at this?
By: Mark Spencer (markster) 2004-03-02 18:46:46.000-0600

Fixed in CVS
By: Digium Subversion (svnbot) 2008-01-15 14:45:22.000-0600

Repository: asterisk
Revision: 2301

U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r2301 | markster | 2008-01-15 14:45:22 -0600 (Tue, 15 Jan 2008) | 2 lines

Process SDP on 183 session progress (bug ASTERISK-924)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=2301
By: Digium Subversion (svnbot) 2008-01-15 14:45:23.000-0600

Repository: asterisk
Revision: 2302

U   branches/v1-0_stable/channels/chan_sip.c

------------------------------------------------------------------------
r2302 | markster | 2008-01-15 14:45:23 -0600 (Tue, 15 Jan 2008) | 2 lines

Process SDP on 183 session progress (bug ASTERISK-924)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=2302