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DTMF missing in outbound calls, specific scenario

Marek.Jiricek
Level 1
Level 1

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

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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View solution in original post

12 Replies 12

Ayodeji Okanlawon
VIP Alumni
VIP Alumni

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..

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

Please rate all useful posts

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.

ok..please send

debug voip rtp session named-event

debug ccsip messages

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

Please rate all useful posts

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.

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

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

Please remove this command

dial-peer voice 3 voip

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

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

Please rate all useful posts

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

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

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

Please rate all useful posts

Nick Richards
Level 1
Level 1

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

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.

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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