DTMF does not work from PSTN(cellphone or your home phone)
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 kill or resolve or delete all DTMF signals on sofia sip?! Is problem from some configuration in sofia?! any idea?
Check what method your SIP provider uses for DTMF: is it RFC2833, in-band or INFO? Then make sure you have the same set in SIP profile. If it does not help, capture wireshark from SIP provider and debug log from FreeSWITCH. It will give you all you need to pinpoint the problem. On 13-03-19 02:30 PM, Sayyed Mohammad Emami Razavi 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 kill or resolve or delete all DTMF signals on sofia sip?! Is problem from some configuration in sofia?! any idea?
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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
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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participants (3)
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Matt Broad
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Sayyed Mohammad Emami Razavi
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Victor Chukalovskiy