[cisco-voip] R: R: DTMF outbound Issue un CUBE
Somphol Boonjing
somphol at gmail.com
Sun Sep 29 17:39:53 EDT 2013
Hi Alessandro,
The IP phone should support 'rtp-nte' and the SIP request offers "rtp-ntp".
As Divin has pointed out the CUCM doesn't accept 'rtp-nte' offer in its
reply to CUBE's INVITE (with rtp-nte in SDP).
Given that MTP is not required at all in this scenarios (because both sides
can do 'rtp-nte'), that leaves one point the DTMF Signaling Method settings
on the SIP Trunk. The safe option is "No Preference" according to Cisco
doc.
(REF: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html
)
Over Unified CM SIP trunks, Cisco recommends configuring the DTMF Signaling
Method to *No Preference*. This setting allows Unified CM to make an
optimal decision for DTMF and to minimize MTP allocation.
I suspect it could be set to out-of-band (OOB).
Regards,
--Somphol.
On Sun, Sep 29, 2013 at 9:09 PM, Alessandro Bertacco <
bertacco.alessandro at alice.it> wrote:
> Hi Somphol,****
>
> that dial-peer is not involved in the scenario.****
>
> ** **
>
> The dial-peer is for H323 communication that works fine also for DTMF.****
>
> ** **
>
> The two dial peer involved in the scenario are two SIP dial-peer 2000 and
> 2001.****
>
> ** **
>
> Thanks!****
>
> ** **
>
> Alessandro****
>
> ** **
>
> *Da:* cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] *Per conto
> di *Somphol Boonjing
> *Inviato:* domenica 29 settembre 2013 11:21
> *A:* cisco-voip at puck.nether.net
> *Oggetto:* Re: [cisco-voip] R: R: DTMF outbound Issue un CUBE****
>
> ** **
>
> Hi, ****
>
> dtmf-relay on dial-peer voice 5 should be changed to 'rtp-nte'. ****
>
> dial-peer voice 5 voip****
>
> description Voice CCM Imbound****
>
> preference 5****
>
> answer-address 799****
>
> destination-pattern 5..****
>
> progress_ind setup enable 3****
>
> modem passthrough nse codec g711ulaw****
>
> session target ipv4:192.168.97.100****
>
> voice-class codec 1 ****
>
> voice-class h323 1****
>
> dtmf-relay h245-alphanumeric <=== rtp-nte here.****
>
> ** **
>
> Regards,****
>
> ** **
>
> On Sun, Sep 29, 2013 at 6:57 PM, Divin John (dijohn) <dijohn at cisco.com>
> wrote:****
>
> CUCM is dropping rtp-nte in the 200 OK.****
>
> ** **
>
> ** **
>
> Sep 29 10:42:09.684 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:***
> *
>
> Sent: ****
>
> INVITE sip:0142213015 at 192.168.97.100:5060 SIP/2.0****
>
> Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487****
>
> From: "0125011302" <sip:0125011302 at link.voipvoice.it>;tag=3D9E9B60-1C8F***
> *
>
> To: <sip:0142213015 at 192.168.97.100>****
>
> Date: Sun, 29 Sep 2013 08:42:09 GMT****
>
> Call-ID: DCE89023-281911E3-B2879FF4-A027EC8D at 192.168.97.230****
>
> Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat****
>
> Min-SE: 1800****
>
> Cisco-Guid: 3705908658-0672731619-2994839540-2686971021****
>
> User-Agent: Cisco-SIPGateway/IOS-12.x****
>
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
> NOTIFY, INFO, REGISTER****
>
> CSeq: 101 INVITE****
>
> Timestamp: 1380444129****
>
> Contact: <sip:0125011302 at 192.168.97.230:5060>****
>
> History-Info: <sip:0142213015 at 192.168.97.100:5060>;index=1****
>
> Expires: 180****
>
> Allow-Events: telephone-event****
>
> Max-Forwards: 26****
>
> Session-Expires: 1800****
>
> Content-Type: application/sdp****
>
> Content-Disposition: session;handling=required****
>
> Content-Length: 323****
>
> ** **
>
> v=0****
>
> o=anonymous 138044412965 138044412965 IN IP4 109.238.17.166****
>
> s=SIP Call****
>
> c=IN IP4 192.168.97.230****
>
> t=0 0****
>
> m=audio 17050 RTP/AVP 18 0 8 101****
>
> b=AS:26****
>
> a=rtpmap:18 G729/8000/1****
>
> a=fmtp:18 annexb=no****
>
> a=rtpmap:0 PCMU/8000/1****
>
> a=rtpmap:8 PCMA/8000/1****
>
> a=rtpmap:101 telephone-event/8000****
>
> a=fmtp:101 0-15****
>
> a=ptime:20****
>
> a=sendrecv****
>
> ** **
>
> ** **
>
> ** **
>
> Sep 29 10:42:27.113 rome: //13639/DCE3ADB2B281/SIP/Msg/ccsipDisplayMsg:***
> *
>
> Received: ****
>
> SIP/2.0 200 OK****
>
> Via: SIP/2.0/UDP 192.168.97.230:5060;branch=z9hG4bK1DC12487****
>
> From: "0125011302" <sip:0125011302 at link.voipvoice.it>;tag=3D9E9B60-1C8F***
> *
>
> To: <sip:0142213015 at 192.168.97.100
> >;tag=366518~2e6a0f22-2647-4b41-a189-3f8120ac75dc-21992748****
>
> Date: Sun, 29 Sep 2013 08:42:09 GMT****
>
> Call-ID: DCE89023-281911E3-B2879FF4-A027EC8D at 192.168.97.230****
>
> CSeq: 101 INVITE****
>
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY****
>
> Allow-Events: presence****
>
> Supported: replaces****
>
> Supported: X-cisco-srtp-fallback****
>
> Supported: Geolocation****
>
> Session-Expires: 1800;refresher=uas****
>
> Require: timer****
>
> P-Asserted-Identity: <sip:555 at 192.168.97.100>****
>
> Remote-Party-ID: <sip:555 at 192.168.97.100
> >;party=called;screen=yes;privacy=off****
>
> Contact: <sip:0142213015 at 192.168.97.100:5060>****
>
> Content-Type: application/sdp****
>
> Content-Length: 193****
>
> ** **
>
> v=0****
>
> o=CiscoSystemsCCM-SIP 366518 1 IN IP4 192.168.97.100****
>
> s=SIP Call****
>
> c=IN IP4 192.168.97.50****
>
> b=TIAS:64000****
>
> b=AS:64****
>
> t=0 0****
>
> m=audio 11078 RTP/AVP 0 ===> 101 is missing****
>
> b=AS:26****
>
> a=rtpmap:0 PCMU/8000****
>
> a=ptime:20****
>
> ** **
>
> Regards,****
>
> Divin****
>
> *From: *Alessandro Bertacco <bertacco.alessandro at alice.it>
> *Date: *Sunday, 29 September 2013 11:50 AM
> *To: *"cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
> *Cc: *Divin John <dijohn at cisco.com>
> *Subject: *R: R: [cisco-voip] DTMF outbound Issue un CUBE****
>
> ** **
>
> Hi all,****
>
> here the debug of an incoming calls, but the same issue happen in the
> outbound call.****
>
> ****
>
> Thank you regards.****
>
> ****
>
> Alessandro****
>
> ****
>
> *Da:* Divin John (dijohn) [mailto:dijohn at cisco.com <dijohn at cisco.com>]
> *Inviato:* sabato 28 settembre 2013 19:44
> *A:* Alessandro Bertacco; cisco-voip at puck.nether.net
> *Oggetto:* Re: R: [cisco-voip] DTMF outbound Issue un CUBE****
>
> ****
>
> Hi Alessandro,****
>
> ****
>
> Try calling ****
>
> http://www.testcall.com/222-1111.html****
>
> ****
>
> *Or an iVR to check.*****
>
> ****
>
> *Regards,*****
>
> *Divin*****
>
> ****
>
> *From: *Alessandro Bertacco <bertacco.alessandro at alice.it>
> *Date: *Saturday, 28 September 2013 6:04 PM
> *To: *Divin John <dijohn at cisco.com>, "cisco-voip at puck.nether.net" <
> cisco-voip at puck.nether.net>
> *Subject: *R: [cisco-voip] DTMF outbound Issue un CUBE****
>
> ****
>
> Hi John,****
>
> thank you for the answer.****
>
> Yes, I mean RFC-2833/rtp-nte.****
>
> ****
>
> At the moment I don't have the debug, but on Monday I'll capture ccsip
> mess from the cube and I'll post it.****
>
> Thank you.****
>
> ****
>
> Regards****
>
> Alessandro****
>
> ****
>
> ****
>
> ****
>
> *Da:* Divin John (dijohn) [mailto:dijohn at cisco.com <dijohn at cisco.com>]
> *Inviato:* sabato 28 settembre 2013 16:42
> *A:* Alessandro Bertacco; cisco-voip at puck.nether.net
> *Oggetto:* Re: [cisco-voip] DTMF outbound Issue un CUBE****
>
> ****
>
> Hey Alessandro,****
>
> ****
>
> What do mean by in band? ****
>
> ****
>
> Inband RFC-2833/rtp-nte****
>
> ****
>
> Or****
>
> ****
>
> Inband PCM Audio (Raw tones in the G711 stream)****
>
> ****
>
> Do you have a debug ccsip mess from the CUBE?****
>
> ****
>
> Regards,****
>
> Divin****
>
> ****
>
> *From: *Alessandro Bertacco <bertacco.alessandro at alice.it>
> *Date: *Saturday, 28 September 2013 5:22 PM
> *To: *"cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
> *Subject: *[cisco-voip] DTMF outbound Issue un CUBE****
>
> ****
>
> Hi all,****
>
> I've a problem with DTMF relay only on outbound direction in this
> topology scenario:****
>
> ****
>
> SIP ISP <---SIP--->Cisco UBE ISR 2811 <--SIP---> Cisco UCM 8.6.X
> <---SCCP---> IP Phone ****
>
> ****
>
> I'm using codec g711 with the provider that tell me to use in-band DTMF.**
> **
>
> ****
>
> SIP trunk in CUCM 8.6 configured as standard without MTP flag, outbound
> and inbound calls works fine, but DTMF relay works only from SIP provider
> to Cisco IP Phone.****
>
> ****
>
> ****
>
> From inside, any DTMF tone are relayed to the SIP provider.****
>
> ****
>
> For example, call from SCCP Phone to my mobile phone, from mobile phone I
> can't hear any DTMF composed to the SCCP phone, but DTMF composed from the
> mobil ephone are recived to the internal SCCP phone.****
>
> ****
>
> This happen always the same in that two scenario:****
>
> Mobile phone ------> SIP ISP ---SIP--->Cisco UBE ISR 2811 --SIP---> Cisco
> UCM 8.6.X ---SCCP---> IP Phone****
>
> IP Phone ---SCCP---> Cisco UCM 8.6.X ---SIP---Cisco UBE ISR 2811 ---SIP-->
> SIP ISP ----> Mobile Phone****
>
> ****
>
> Can anyone help me?****
>
> ****
>
> Thank you very much****
>
> ****
>
> Regards****
>
> ****
>
> Alessandro Bertacco****
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ** **
>
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