[asterisk-bugs] [Asterisk 0010862]: Problem with DTMF being passed from Cisco GW to asterisk on ingress calls Description

[noreply at bugs.digium.com]

The following issue requires your FEEDBACK. 
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http://bugs.digium.com/view.php?id=10862 
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Reported By:                clone
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10862
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.11  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             10-01-2007 05:35 CDT
Last Modified:              10-01-2007 08:18 CDT
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Summary:                    Problem with DTMF being passed from Cisco GW to
asterisk on ingress calls
Description
Description: 
After upgrading to to release 1.4.11, asterisk no longer processes DTMF
from caller. I have my VOIP peer setup to pass info via RFC2833. This
affects applications such as DISA, and Menus etc..
sip.conf:
[993]
type = friend
host = 85.28.xxx.xxx
canreinvite = no
context = geronimo
disallow = all
allow = g729,gsm,ulaw,alaw
;dtmfmode = info
dtmfmode = rfc2833
;dtmfmode = auto
fromdomain = 85.28.xxx.xxx
insecure = very
callerid = "993"
t38pt_udptl = yes
nat = no

cisco.conf (AS5350):

dial-peer voice 993 voip
 destination-pattern 993
 voice-class codec 1
 session protocol sipv2
 session target ipv4:80.89.xxx.xxx
 session transport udp
 dtmf-relay rtp-nte
 no vad

sip show peer 993:


  * Name       : 993
  Realtime peer: No
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Context      : geronimo
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  FromDomain   : 85.28.xxx.xxx
  Callgroup    :
  Pickupgroup  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : No
  Callerid     : "993" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : port,invite
  Nat          : RFC3581
  ACL          : No
  T38 pt UDPTL : Yes
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : 85.28.xxx.xxx
  Addr->IP     : 85.28.xxx.xxx Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Def. Username:
  SIP Options  : (none)
  Codecs       : 0x10e (gsm|ulaw|alaw|g729)
  Codec Order  : (g729:20,gsm:20,ulaw:20,alaw:20)
  Auto-Framing:  No
  Status       : Unmonitored
  Useragent    :
  Reg. Contact :

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---------------------------------------------------------------------- 
 file - 10-01-07 08:18  
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Please provide complete console output with DTMF logging enabled (dtmf in
logger.conf), complete sip debug, and a complete rtp debug. 

Issue History 
Date Modified   Username       Field                    Change               
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10-01-07 08:18  file           Note Added: 0071226                          
10-01-07 08:18  file           Status                   new => feedback     
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