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Summary:ASTERISK-08576: Outbound DTMF Fails with Sipgate using RFC2833
Reporter:law1213 (law1213)Labels:
Date Opened:2007-01-14 09:31:22.000-0600Date Closed:2011-06-07 14:02:46
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:When using call files for asterisk outgoing calls, only the first DTMF tone is detected all others are ignored. Problem only occurs for outgoing calls using call files, incoming calls dtmf tones are detected correctly, outgoing calls (using DIAL command) have their dtmf tones detected correctly also.

Same problem occurs using ulaw alaw with inband, have also tried info dtmfmodes. Problem occurs in Asterisk 1.2.1 and 1.4.

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

sip.conf for sipgate
--------------------
disable=all
allow=gsm
allow=alaw
allow=alaw
allow=ulaw
allow=g729
;allow=gsm
allow=slinear
allow=speex
allow=iLBC
dtmftone=rfc2833

[sipgate-out]
type=friend
insecure=very ; otherwise I get authentication errors
nat=no
username=****
fromuser=****
fromdomain=sipgate.co.uk
secret=****
host=sipgate.co.uk
qualify=no
dtmfmode=rfc2833

DIAL PLAN EXTRACT
-----------------
;Callback give number
exten => 2738876,1,Authenticate(7629)
exten => 2738876,2,NoOp(Incoming call)
exten => 2738876,3,Read(IDNUM|enter-phone-number10|0)
exten => 2738876,4,SayDigits(${IDNUM})
exten => 2738876,5,Playback(goodbye)
exten => 2738876,6,Hangup()
exten => h,1,System(echo channel: ${OUTBOUNDTRUNK}/${IDNUM} > /tmp/${IDNUM})
exten => h,2,System(echo context: incoming >> /tmp/${IDNUM})
exten => h,3,System(echo extension: ${IDNUM} >> /tmp/${IDNUM})
exten => h,4,System(echo priority: 1 >> /tmp/${IDNUM})
exten => h,5,System(echo callerid: **** >> /tmp/${IDNUM})
exten => h,6,System(echo sleep 30 > /tmp/${IDNUM}.2)
exten => h,7,System(echo cp /tmp/${IDNUM} /var/spool/asterisk/outgoing >> /tmp/$
exten => h,8,System(chmod 775 /tmp/${IDNUM}.2)
exten => h,9,System(/tmp/${IDNUM}.2)
exten => h,10,Hangup()



DEBUG Info from SIP DEBUG
-------------------------
Scheduling destruction of call '6ea29a424f2ab92409f973e26d303050@192.168.1.9' in 32000 ms
Jan 14 15:10:44 NOTICE[8144]: chan_sip.c:9661 handle_response_register: Outbound Registration: Expiry for sipgate.co.uk is 120 sec (Scheduling reregistration in 105 s)
prometheus*CLI>
<-- SIP read from 217.10.79.23:5060:
INFO sip:****@****:5060 SIP/2.0
Record-Route: <sip:217.10.79.23;lr=on>
Record-Route: <sip:217.10.79.8;ftag=as71572e19;lr=on>
Via: SIP/2.0/UDP 217.10.79.23;branch=z9hG4bK6fd1.a0db84b7.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK6fd1.c17348e7.0
Via: SIP/2.0/UDP 217.10.69.7:5060;branch=z9hG4bK3c59252f;rport=5060
From: <sip:****@sipgate.co.uk>;tag=as71572e19
To: "919502" <sip:****@sipgate.co.uk>;tag=as0152eac5
Contact: <sip:****@217.10.69.7>
Call-ID: 2f56fb8074efb0651e012788702cda58@sipgate.co.uk
CSeq: 102 INFO
User-Agent: sipgate asterisk
Max-Forwards: 15
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=1
Duration=250

--- (15 headers 2 lines)---
Receiving INFO!
* DTMF-relay event received: 1
Transmitting (no NAT) to 217.10.79.23:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.23;branch=z9hG4bK6fd1.a0db84b7.0;received=217.10.79.23
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK6fd1.c17348e7.0
Via: SIP/2.0/UDP 217.10.69.7:5060;branch=z9hG4bK3c59252f;rport=5060
Record-Route: <sip:217.10.79.23;lr=on>
Record-Route: <sip:217.10.79.8;ftag=as71572e19;lr=on>
From: <sip:****@sipgate.co.uk>;tag=as71572e19
To: "919502" <sip:****@sipgate.co.uk>;tag=as0152eac5
Call-ID: 2f56fb8074efb0651e012788702cda58@sipgate.co.uk
CSeq: 102 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:****@192.168.1.11>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
Destroying call '6ea29a424f2ab92409f973e26d303050@192.168.1.9'
prometheus*CLI>
<-- SIP read from 217.10.79.23:5060:

--- (0 headers 0 lines) Nat keepalive ---
Comments:By: Tilghman Lesher (tilghman) 2007-01-14 22:02:43.000-0600

Your SIP debug appears to be showing SIP INFO DTMF, instead of RFC2833 DTMF.  These are two different settings.  Are you sure your sipgate device supports RFC2833 or that you have that setting currently turned on?
By: Joshua C. Colp (jcolp) 2007-01-15 11:09:39.000-0600

rtp debug is also needed if RFC2833 is involved.
By: law1213 (law1213) 2007-01-18 06:11:33.000-0600

Thank you for pointing this out I will investigate and provide you with more info. I have email sipgate to find out if they support rfc2833. Any idea why the dtmfmode might be ignored? Incoming calls via sipgate respond normally to dtmf tones and I think (but will confirm) outgoing calls made with sip phone via asterisk and sipgate respond normally to dtmf tones on both the caller and callee side. It seems only when using call files this problem occurs.
By: Serge Vecher (serge-v) 2007-01-25 13:18:58.000-0600

there was an issue with SIP INFO dtmf recently fixed in 8597. Can you please try revision > 51311 and see if that works?
By: Russell Bryant (russell) 2007-01-27 00:38:47.000-0600

I'm closing this out since it is a configuration issue.