<div dir="ltr">Hi <span style="font-family:arial,sans-serif;font-size:13px;white-space:nowrap">Sayyed,</span><div><span style="font-family:arial,sans-serif;font-size:13px;white-space:nowrap"><br></span></div><div style><font face="arial, sans-serif"><span style="white-space:nowrap">I think to capture the DTMF packets you will need to remove the port 5060 from your tcpdump. </span></font></div>
<div style><font face="arial, sans-serif"><span style="white-space:nowrap">You should then lookout for the RTP events in the log, wireshark is a great tool for detemining issues like this.</span></font></div><div style><font face="arial, sans-serif"><span style="white-space:nowrap"><br>
</span></font></div><div style><font face="arial, sans-serif"><span style="white-space:nowrap">Also, as suggested by Victor, have you checked with your SIP provider which method they use to send the DTMF? </span></font></div>
<div style><font face="arial, sans-serif"><span style="white-space:nowrap">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.</span></font></div>
<div><span style="font-family:arial,sans-serif;font-size:13px;white-space:nowrap"><br></span></div><div style><span style="font-family:arial,sans-serif;font-size:13px;white-space:nowrap">thanks</span></div><div style><span style="font-family:arial,sans-serif;font-size:13px;white-space:nowrap">Matt</span></div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On 26 March 2013 04:49, Sayyed Mohammad Emami Razavi <span dir="ltr"><<a href="mailto:emamirazavi@gmail.com" target="_blank">emamirazavi@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="gmail_quote"><div class="im">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,<br>
what is the problem? This problem is new and beforehand i had no problem with dtmf from cellphone!<br>
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.<br></div>May my trunk filter, kill, resolve or delete all DTMF or kpml signals on sofia sip?!<div class="im">
<br>
Is problem from some configuration in sofia?! any idea?<br></div>my provider(trunk and ...) uses cisco IPPBX and i use FS certainly.<br><br><font size="4">$ tcpdump -nq -s 0 -A -vvv -i eth1 port 5060</font><br><br><i><b>tcpdump from local IP phone:</b></i><br>
12:21:57.993515 IP (tos 0x60, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 1185)<br> 192.168.5.2.sip > 192.168.6.10.sip: [udp sum ok] UDP, length 1157<br>E`....@.?..........<br>.......OINVITE <a href="http://sip:288@192.168.6.10:5060" target="_blank">sip:288@192.168.6.10:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK822577c262cc<br>From: "M.Emami" <<a href="mailto:sip%3A334@192.168.5.2" target="_blank">sip:334@192.168.5.2</a>>;tag=425253~ba780a27-1f50-49c4-afac-f9a30086d185-30836414<br>
To: <<a href="mailto:sip%3A288@192.168.6.10" target="_blank">sip:288@192.168.6.10</a>><br>Date: Tue, 26 Mar 2013 04:13:41 GMT<br>Call-ID: <a href="mailto:89560e00-15112075-31ab-205a8c0@192.168.5.2" target="_blank">89560e00-15112075-31ab-205a8c0@192.168.5.2</a><br>
Supported: timer,resource-priority,replaces<br>Min-SE: 1800<br>User-Agent: Cisco-CUCM8.6<br>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY<br>CSeq: 101 INVITE<br>Expires: 180<br>
<u><i><span style="color:rgb(51,102,255)">Allow-Events: presence, kpml</span></i></u><br>
Supported: X-cisco-srtp-fallback<br>Supported: Geolocation<br>Cisco-Guid: 2304118272-0000065536-0000023496-0033925312<br>Session-Expires: 1800<br>P-Asserted-Identity: "M.Emami" <<a href="mailto:sip%3A334@192.168.5.2" target="_blank">sip:334@192.168.5.2</a>><br>
Remote-Party-ID: "M.Emami" <<a href="mailto:sip%3A334@192.168.5.2" target="_blank">sip:334@192.168.5.2</a>>;party=calling;screen=yes;privacy=off<br>Contact: <<a href="http://sip:334@192.168.5.2:5060" target="_blank">sip:334@192.168.5.2:5060</a>><br>
Max-Forwards: 70<br>Content-Type: application/sdp<br>Content-Length: 212<br><br>v=0<br>o=CiscoSystemsCCM-SIP 425253 1 IN IP4 192.168.5.2<br>s=SIP Call<br>c=IN IP4 192.168.5.2<br>t=0 0<br>m=audio 27762 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>
a=ptime:20<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br><br>12:21:57.994024 IP (tos 0x0, ttl 64, id 60838, offset 0, flags [none], proto UDP (17), length 361)<br> 192.168.6.10.sip > 192.168.5.2.sip: [bad udp cksum 4278!] UDP, length 333<br>
E..i....@......<br>.........U..SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK822577c262cc<br>From: "M.Emami" <<a href="mailto:sip%3A334@192.168.5.2" target="_blank">sip:334@192.168.5.2</a>>;tag=425253~ba780a27-1f50-49c4-afac-f9a30086d185-30836414<br>
To: <<a href="mailto:sip%3A288@192.168.6.10" target="_blank">sip:288@192.168.6.10</a>><br>Call-ID: <a href="mailto:89560e00-15112075-31ab-205a8c0@192.168.5.2" target="_blank">89560e00-15112075-31ab-205a8c0@192.168.5.2</a><br>
CSeq: 101 INVITE<br>User-Agent: Configured by 2600hz!<br>
Content-Length: 0<br><br><br><b><i>tcpdump from outbound cell phone or any outbound calls:</i></b><br>12:24:21.596280 IP (tos 0x60, ttl 63, id 0, offset 0, flags [DF], proto UDP (17), length 1180)<br> 192.168.5.2.sip > 192.168.6.10.sip: [udp sum ok] UDP, length 1152<br>
E`....@.?..........<br>........INVITE <a href="http://sip:288@192.168.6.10:5060" target="_blank">sip:288@192.168.6.10:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK8230c25bb59<br>From: <<a href="mailto:sip%3A09102260264@192.168.5.2" target="_blank">sip:09102260264@192.168.5.2</a>>;tag=425297~ba780a27-1f50-49c4-afac-f9a30086d185-30836444<br>
To: <<a href="mailto:sip%3A288@192.168.6.10" target="_blank">sip:288@192.168.6.10</a>><br>Date: Tue, 26 Mar 2013 04:16:04 GMT<br>Call-ID: <a href="mailto:de921f80-15112104-31b0-205a8c0@192.168.5.2" target="_blank">de921f80-15112104-31b0-205a8c0@192.168.5.2</a><br>
Supported: timer,resource-priority,replaces<br>Min-SE: 1800<br>User-Agent: Cisco-CUCM8.6<br>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY<br>CSeq: 101 INVITE<br>Expires: 180<br>
<u><i><span style="color:rgb(51,102,255)">Allow-Events: presence</span></i></u><br>
Supported: X-cisco-srtp-fallback<br>Supported: Geolocation<br>Cisco-Guid: 3734118272-0000065536-0000023506-0033925312<br>Session-Expires: 1800<br>P-Asserted-Identity: <<a href="mailto:sip%3A09102260264@192.168.5.2" target="_blank">sip:09102260264@192.168.5.2</a>><br>
Remote-Party-ID: <<a href="mailto:sip%3A09102260264@192.168.5.2" target="_blank">sip:09102260264@192.168.5.2</a>>;party=calling;screen=yes;privacy=off<br>Contact: <<a href="http://sip:09102260264@192.168.5.2:5060" target="_blank">sip:09102260264@192.168.5.2:5060</a>><br>
Max-Forwards: 69<br>Content-Type: application/sdp<br>Content-Length: 212<br><br>v=0<br>o=CiscoSystemsCCM-SIP 425297 1 IN IP4 192.168.5.2<br>s=SIP Call<br>c=IN IP4 192.168.5.2<br>t=0 0<br>m=audio 27790 RTP/AVP 0 101<br>a=rtpmap:0 PCMU/8000<br>
a=ptime:20<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br><br>12:24:21.596789 IP (tos 0x0, ttl 64, id 60847, offset 0, flags [none], proto UDP (17), length 358)<br> 192.168.6.10.sip > 192.168.5.2.sip: [bad udp cksum c03a!] UDP, length 330<br>
E..f....@..z...<br>.........R..SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.5.2:5060;branch=z9hG4bK8230c25bb59<br>From: <<a href="mailto:sip%3A09102260264@192.168.5.2" target="_blank">sip:09102260264@192.168.5.2</a>>;tag=425297~ba780a27-1f50-49c4-afac-f9a30086d185-30836444<br>
To: <<a href="mailto:sip%3A288@192.168.6.10" target="_blank">sip:288@192.168.6.10</a>><br>Call-ID: <a href="mailto:de921f80-15112104-31b0-205a8c0@192.168.5.2" target="_blank">de921f80-15112104-31b0-205a8c0@192.168.5.2</a><br>
CSeq: 101 INVITE<br>User-Agent: Configured by 2600hz!<br>
Content-Length: 0<br><br>
</div><br>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Thanks<br>Matt<br><br>This email and any attachments to it are confidential and are intended solely for the use of the individual to whom it is addressed. Any views or opinions expressed are solely those of the author and do not necessarily represent those of InverOak Limited.<br>
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