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Summary:ASTERISK-10423: Problem with DTMF being passed from Cisco GW to asterisk on ingress calls Description
Reporter:Igor Khoroshev (clone)Labels:
Date Opened:2007-10-01 05:35:59Date Closed:2011-06-07 14:03:23
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) asterisk.log
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 :
Comments:By: Joshua C. Colp (jcolp) 2007-10-01 08:18:15

Please provide complete console output with DTMF logging enabled (dtmf in logger.conf), complete sip debug, and a complete rtp debug.
By: Igor Khoroshev (clone) 2007-10-01 23:14:58

Sip and rtp debug in the attached file asterisk.log.
dtmf.log is empty.


By: Joshua C. Colp (jcolp) 2007-10-02 08:48:56

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?
By: Igor Khoroshev (clone) 2007-10-02 21:24:43

Thank you. It work.
By: Joshua C. Colp (jcolp) 2007-10-03 09:16:51

Closed, configuration issue.