[cisco-voip] R: R: R: DTMF outbound Issue un CUBE

Wed Oct 2 15:11:55 EDT 2013

Hi Somphol.

   Thank you for the answer, but the dial-peer you suggest don't solve my
issue.

I need to investigate CUCM trace and if possible evaluate IOS upgrade on the
CUBE.

Here the SIP trunk configuration snapshot from the cucm side.

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
<mailto:sip%3A0125011302 at link.voipvoice.it> >;tag=3D9E9B60-1C8F

To: <sip:0142213015 at 192.168.97.100 <mailto:sip%3A0142213015 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 <tel:18%200%208%20101> 

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
<mailto:sip%3A0125011302 at link.voipvoice.it> >;tag=3D9E9B60-1C8F

To: <sip:0142213015 at 192.168.97.100 <mailto:sip%3A0142213015 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
<mailto:sip%3A555 at 192.168.97.100> >

Remote-Party-ID: <sip:555 at 192.168.97.100 <mailto:sip%3A555 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] 
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] 
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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20131002/0057882b/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.jpg
Type: application/octet-stream
Size: 85359 bytes
Desc: not available
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20131002/0057882b/attachment.obj>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image002.jpg
Type: application/octet-stream
Size: 105475 bytes
Desc: not available
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20131002/0057882b/attachment-0001.obj>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image003.jpg
Type: application/octet-stream
Size: 98133 bytes
Desc: not available
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20131002/0057882b/attachment-0002.obj>