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

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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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:              10-02-2007 08:48 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 - 10-02-07 08:48  
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This INVITE matched peer entry 30800, not the sip entry you gave in the
initial bug note. It does not appear as though that peer is configured for
rfc2833, thus it was not negotiated. If you change it to dtmfmode=rfc2833
does it work? 

Issue History 
Date Modified   Username       Field                    Change               
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10-02-07 08:48  file           Note Added: 0071306                          
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