[Freeswitch-users] DTMF does not work from PSTN(cellphone or your home phone)

Matt Broad matt at inveroak.com
Tue Mar 26 11:41:19 MSK 2013


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 at 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`.... at .?..........
> .......OINVITE sip:288 at 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 at 192.168.5.2
> >;tag=425253~ba780a27-1f50-49c4-afac-f9a30086d185-30836414
> To: <sip:288 at 192.168.6.10>
> Date: Tue, 26 Mar 2013 04:13:41 GMT
> Call-ID: 89560e00-15112075-31ab-205a8c0 at 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 at 192.168.5.2>
> Remote-Party-ID: "M.Emami" <sip:334 at 192.168.5.2
> >;party=calling;screen=yes;privacy=off
> Contact: <sip:334 at 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.... at ......
> .........U..SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK822577c262cc
> From: "M.Emami" <sip:334 at 192.168.5.2
> >;tag=425253~ba780a27-1f50-49c4-afac-f9a30086d185-30836414
> To: <sip:288 at 192.168.6.10>
> Call-ID: 89560e00-15112075-31ab-205a8c0 at 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`.... at .?..........
> ........INVITE sip:288 at 192.168.6.10:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK8230c25bb59
> From: <sip:09102260264 at 192.168.5.2
> >;tag=425297~ba780a27-1f50-49c4-afac-f9a30086d185-30836444
> To: <sip:288 at 192.168.6.10>
> Date: Tue, 26 Mar 2013 04:16:04 GMT
> Call-ID: de921f80-15112104-31b0-205a8c0 at 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 at 192.168.5.2>
> Remote-Party-ID: <sip:09102260264 at 192.168.5.2
> >;party=calling;screen=yes;privacy=off
> Contact: <sip:09102260264 at 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.... at ..z...
> .........R..SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK8230c25bb59
> From: <sip:09102260264 at 192.168.5.2
> >;tag=425297~ba780a27-1f50-49c4-afac-f9a30086d185-30836444
> To: <sip:288 at 192.168.6.10>
> Call-ID: de921f80-15112104-31b0-205a8c0 at 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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