Hi Sayyed,

I think to capture the DTMF packets you will need to remove the port 5060 from your tcpdump.  
You should then lookout for the RTP events in the log, wireshark is a great tool for detemining issues like this.

Also, as suggested by Victor, have you checked with your SIP provider which method they use to send the DTMF?  
If they are sending inband and you have not specifically set to listen to inband in the dialplan (start_dtmf), then Freeswitch will ignore them.

thanks
Matt


On 26 March 2013 04:49, Sayyed Mohammad Emami Razavi <emamirazavi@gmail.com> wrote:
No digits are passed from cellphone or your home phone but when you press any key from your registered IP phone on local lan, digits are passed very well,
what is the problem? This problem is new and beforehand i had no problem with dtmf from cellphone!
Every thing is good, all logs are good, my IVR that gets digits and transfer calls are good and no problem exists in these aspects.
May my trunk filter, kill, resolve or delete all DTMF or kpml signals on sofia sip?!

Is problem from some configuration in sofia?! any idea?
my provider(trunk and ...) uses cisco IPPBX and i use FS certainly.

$ tcpdump -nq -s 0 -A -vvv -i eth1 port 5060

tcpdump from local IP phone:
12:21:57.993515 IP (tos 0x60, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 1185)
    192.168.5.2.sip > 192.168.6.10.sip: [udp sum ok] UDP, length 1157
E`....@.?..........
.......OINVITE sip:288@192.168.6.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK822577c262cc
From: "M.Emami" <sip:334@192.168.5.2>;tag=425253~ba780a27-1f50-49c4-afac-f9a30086d185-30836414
To: <sip:288@192.168.6.10>
Date: Tue, 26 Mar 2013 04:13:41 GMT
Call-ID: 89560e00-15112075-31ab-205a8c0@192.168.5.2
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 2304118272-0000065536-0000023496-0033925312
Session-Expires:  1800
P-Asserted-Identity: "M.Emami" <sip:334@192.168.5.2>
Remote-Party-ID: "M.Emami" <sip:334@192.168.5.2>;party=calling;screen=yes;privacy=off
Contact: <sip:334@192.168.5.2:5060>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 212

v=0
o=CiscoSystemsCCM-SIP 425253 1 IN IP4 192.168.5.2
s=SIP Call
c=IN IP4 192.168.5.2
t=0 0
m=audio 27762 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

12:21:57.994024 IP (tos 0x0, ttl 64, id 60838, offset 0, flags [none], proto UDP (17), length 361)
    192.168.6.10.sip > 192.168.5.2.sip: [bad udp cksum 4278!] UDP, length 333
E..i....@......
.........U..SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK822577c262cc
From: "M.Emami" <sip:334@192.168.5.2>;tag=425253~ba780a27-1f50-49c4-afac-f9a30086d185-30836414
To: <sip:288@192.168.6.10>
Call-ID: 89560e00-15112075-31ab-205a8c0@192.168.5.2
CSeq: 101 INVITE
User-Agent: Configured by 2600hz!
Content-Length: 0


tcpdump from outbound cell phone or any outbound calls:
12:24:21.596280 IP (tos 0x60, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 1180)
    192.168.5.2.sip > 192.168.6.10.sip: [udp sum ok] UDP, length 1152
E`....@.?..........
........INVITE sip:288@192.168.6.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK8230c25bb59
From: <sip:09102260264@192.168.5.2>;tag=425297~ba780a27-1f50-49c4-afac-f9a30086d185-30836444
To: <sip:288@192.168.6.10>
Date: Tue, 26 Mar 2013 04:16:04 GMT
Call-ID: de921f80-15112104-31b0-205a8c0@192.168.5.2
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 3734118272-0000065536-0000023506-0033925312
Session-Expires:  1800
P-Asserted-Identity: <sip:09102260264@192.168.5.2>
Remote-Party-ID: <sip:09102260264@192.168.5.2>;party=calling;screen=yes;privacy=off
Contact: <sip:09102260264@192.168.5.2:5060>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 212

v=0
o=CiscoSystemsCCM-SIP 425297 1 IN IP4 192.168.5.2
s=SIP Call
c=IN IP4 192.168.5.2
t=0 0
m=audio 27790 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

12:24:21.596789 IP (tos 0x0, ttl 64, id 60847, offset 0, flags [none], proto UDP (17), length 358)
    192.168.6.10.sip > 192.168.5.2.sip: [bad udp cksum c03a!] UDP, length 330
E..f....@..z...
.........R..SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK8230c25bb59
From: <sip:09102260264@192.168.5.2>;tag=425297~ba780a27-1f50-49c4-afac-f9a30086d185-30836444
To: <sip:288@192.168.6.10>
Call-ID: de921f80-15112104-31b0-205a8c0@192.168.5.2
CSeq: 101 INVITE
User-Agent: Configured by 2600hz!
Content-Length: 0



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--
Thanks
Matt

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