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Beginner

DTMF missing in outbound calls, specific scenario

Hello all,

i'll appreciate any help with my DTMF issue, here is quick overview:

Cisco IP Phone 7941 SCCP/SIP -> CUCM 8.6 -> GW Router 2801 (ipipgw 12.4(24)T8) -> ASA FW -> SIP Trunk provider.

SIP Provider is using DTMF SIP INFO method (inband-voice in SIP signalization).

For inbound calls dtmf works fine, problem is with an outbound calls (initiating call in-to-out) on 7941. I was testing 7940 SIP and it worked fine in both directions, but it needs to be working on 7941 as well as on 7940.

I did a lot of troubleshooting and configuring about this kind of behaviour, but nothing solved this issue.

I guess I can provide you packet captures on CUCM, GW for using with SW Wireshark and a debug on GW. Whatever will help us to find the source and solution of this issue.

Thanks a lot for any advice and ideas while also apologize for non perfect english.

With best regards

Mark

-----

In the attach I give you last output from "debug ccsip mess..." command on R2801 (both non/working DTMF on 7941/7940 SIP) and its probably very common config (one of many setup).

1 ACCEPTED SOLUTION

Accepted Solutions
VIP Mentor

Re: DTMF missing in outbound calls, specific scenario

From the logs..I can see that the invite sent out to ITSP does not cotain any DTMF attributes...

Sent:
INVITE sip:800123456@188.175.113.182:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK18911A

From: "Marek JIRICEK" <226805813>;tag=433A9D98-20C

To: <800123456>

Date: Mon, 08 Apr 2013 15:46:30 GMT

Call-ID: 509B2AE6-9F9A11E2-9377CDA1-7BDD1AAB@192.168.105.20

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 2902615296-65536-460-359246016

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1365435990

Contact: <226805813>

Expires: 180

Allow-Events: kpml, telephone-event

Max-Forwards: 68

Session-Expires:  1800

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 197

v=0

o=CiscoSystemsSIP-GW-UserAgent 3604 6815 IN IP4 192.168.105.20

s=SIP Call

c=IN IP4 192.168.105.20

t=0 0

m=audio 17736 RTP/AVP 8

c=IN IP4 192.168.105.20

a=rtpmap:8 PCMA/8000

a=ptime:20

---------------------------------------------------------no DTMF attribute

Look below and see the invite from CUCM it has

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

which are the dtmf attributes for rtp-nte

Received:
INVITE sip:800123456@192.168.105.20:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.105.21:5060;branch=z9hG4bK1f9641bc9a8cc

From: "Marek JIRICEK" <13>;tag=136424~9c01f162-7d88-4d2e-927f-31024c52083a-27224612

To: <800123456>

Date: Mon, 08 Apr 2013 15:04:59 GMT

Call-ID: ad026500-1621dc9b-1f257-1569a8c0@192.168.105.21

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

Call-Info: <192.168.105.21:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Cisco-Guid: 2902615296-0000065536-0000000460-0359246016

Session-Expires:  1800

P-Asserted-Identity: "Marek JIRICEK" <13>

Remote-Party-ID: "Marek JIRICEK" <13>;party=calling;screen=yes;privacy=off

Contact: <13>

Max-Forwards: 69

Content-Type: application/sdp

Content-Length: 337

v=0

o=CiscoSystemsCCM-SIP 136424 1 IN IP4 192.168.105.21

s=SIP Call

c=IN IP4 192.168.106.214

b=TIAS:64000

b=AS:64

t=0 0

m=audio 20790 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:8 PCMA/8000

a=ptime:20

a=rtpmap:18 G729/8000

a=ptime:20

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

You should check the dial-peer that is matched for this call and ensure you have the proper dtmf type on it

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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12 REPLIES 12
VIP Mentor

DTMF missing in outbound calls, specific scenario

Please configure this..

dial-peer voice 3 voip

dtmf-relay rtp-nte sip-kpml

dial-peer voice 10 voip

dtmf-relay rtp-nte sip-kpml

test again..

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts
Beginner

DTMF missing in outbound calls, specific scenario

Thanks for quick reply, but I've already tested more dtmf-relay setups in dial-peer. Now no change with that either, so post actual outputs isn't necessary.

M.

VIP Mentor

DTMF missing in outbound calls, specific scenario

ok..please send

debug voip rtp session named-event

debug ccsip messages

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts
Beginner

Re: DTMF missing in outbound calls, specific scenario

RTP event from actually working 7940:

1w6d:          s=VoIP d=DSP payload 0x65 ssrc 0xE155D728 sequence 0x33E1 timestamp 0x1                                                                            FAC70

1w6d:  << Pt:101    Evt:1       Pkt:0A 00 00

1w6d:          s=DSP d=VoIP payload 0x65 ssrc 0xE155D728 sequence 0x33E1 timestamp 0x1                                                                            FAC70

1w6d:          Pt:101    Evt:1       Pkt:0A 00 00  >>

1w6d:          s=VoIP d=DSP payload 0x65 ssrc 0xE155D728 sequence 0x33E3 timestamp 0x1                                                                            FAC70

1w6d:  << Pt:101    Evt:1       Pkt:0A 00 A0

1w6d:          s=DSP d=VoIP payload 0x65 ssrc 0xE155D728 sequence 0x33E3 timestamp 0x1                                                                            FAC70

1w6d:          Pt:101    Evt:1       Pkt:0A 00 A0  >>

1w6d:          s=VoIP d=DSP payload 0x65 ssrc 0xE155D728 sequence 0x33E5 timestamp 0x1                                                                            FAC70

...and so on

SIP Messages attached.

Thanks a lot for helping

M.

VIP Mentor

Re: DTMF missing in outbound calls, specific scenario

From the logs..I can see that the invite sent out to ITSP does not cotain any DTMF attributes...

Sent:
INVITE sip:800123456@188.175.113.182:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.105.20:5060;branch=z9hG4bK18911A

From: "Marek JIRICEK" <226805813>;tag=433A9D98-20C

To: <800123456>

Date: Mon, 08 Apr 2013 15:46:30 GMT

Call-ID: 509B2AE6-9F9A11E2-9377CDA1-7BDD1AAB@192.168.105.20

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 2902615296-65536-460-359246016

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1365435990

Contact: <226805813>

Expires: 180

Allow-Events: kpml, telephone-event

Max-Forwards: 68

Session-Expires:  1800

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 197

v=0

o=CiscoSystemsSIP-GW-UserAgent 3604 6815 IN IP4 192.168.105.20

s=SIP Call

c=IN IP4 192.168.105.20

t=0 0

m=audio 17736 RTP/AVP 8

c=IN IP4 192.168.105.20

a=rtpmap:8 PCMA/8000

a=ptime:20

---------------------------------------------------------no DTMF attribute

Look below and see the invite from CUCM it has

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

which are the dtmf attributes for rtp-nte

Received:
INVITE sip:800123456@192.168.105.20:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.105.21:5060;branch=z9hG4bK1f9641bc9a8cc

From: "Marek JIRICEK" <13>;tag=136424~9c01f162-7d88-4d2e-927f-31024c52083a-27224612

To: <800123456>

Date: Mon, 08 Apr 2013 15:04:59 GMT

Call-ID: ad026500-1621dc9b-1f257-1569a8c0@192.168.105.21

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

Call-Info: <192.168.105.21:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Cisco-Guid: 2902615296-0000065536-0000000460-0359246016

Session-Expires:  1800

P-Asserted-Identity: "Marek JIRICEK" <13>

Remote-Party-ID: "Marek JIRICEK" <13>;party=calling;screen=yes;privacy=off

Contact: <13>

Max-Forwards: 69

Content-Type: application/sdp

Content-Length: 337

v=0

o=CiscoSystemsCCM-SIP 136424 1 IN IP4 192.168.105.21

s=SIP Call

c=IN IP4 192.168.106.214

b=TIAS:64000

b=AS:64

t=0 0

m=audio 20790 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:8 PCMA/8000

a=ptime:20

a=rtpmap:18 G729/8000

a=ptime:20

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

You should check the dial-peer that is matched for this call and ensure you have the proper dtmf type on it

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts
Beginner

Re: DTMF missing in outbound calls, specific scenario

I'll check it again, but when i debugged matching dial-peers last time, it was correct...see attach pls to find any mistakes.

Thank you

VIP Mentor

DTMF missing in outbound calls, specific scenario

Please remove this command

dial-peer voice 3 voip

no voice-class sip dtmf-relay force rtp-nte

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts
Beginner

DTMF missing in outbound calls, specific scenario

Problem has been solved - matching for incoming dnis was different, to make it clear:

1w6d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=3

1w6d: //-1/5131ED800000/DPM/dpAssociateIncomingPeerCore:

   Calling Number=13, Called Number=800123456, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

1w6d: //-1/5131ED800000/DPM/dpAssociateIncomingPeerCore:

   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=11

1w6d: //-1/5131ED800000/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=800123456, Peer Info Type=DIALPEER_INFO_SPEECH

I guess it happened because incoming-called command has higher priority than destination-pattern and it just matches that different dial-peer. If u have any ideas in this case, pls contact me. Otherwise I assume this config is working correctly.

Thanks a lot aokanlawon and others

With best regards

Mark

VIP Mentor

DTMF missing in outbound calls, specific scenario

Marek..good to know its fine now..But you havent rated the correct post   and any other usefful post.It helps other

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts
Beginner

DTMF missing in outbound calls, specific scenario

Is that IOS release even supported with 8.6? The Release Notes seem to indicate otherwise:

http://www.cisco.com/en/US/docs/voice_ip_comm/uc_system/UC8.6.1/release_notes/rnipt861.html

As always, whether it's supported and whether it actually works are two entirely different things...

NR

Beginner

DTMF missing in outbound calls, specific scenario

Well, that is always the question. But I dont think that is the correct way of looking on it. How do you explain all correctly working on 7940 with same setup?

M.

Engager

Re: DTMF missing in outbound calls, specific scenario

Hi

for outboumd DTMF issue , please try the below commands

dial-peer voice 3 voip
 description **Outgoing Call to SIP Trunk**
 translation-profile outgoing PSTN_Outgoing
 destination-pattern ..............
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
codec  g711ulaw
no vad !

Thank you

please rate if this weill help

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