04-08-2013 06:56 AM - edited 03-16-2019 04:40 PM
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).
Solved! Go to Solution.
04-08-2013 08:19 AM
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=433A9D98-20C226805813>
To: <800123456>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>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-2722461213>
To: <800123456>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"192.168.105.21:5060>
Cisco-Guid: 2902615296-0000065536-0000000460-0359246016
Session-Expires: 1800
P-Asserted-Identity: "Marek JIRICEK" <13>13>
Remote-Party-ID: "Marek JIRICEK" <13>;party=calling;screen=yes;privacy=off13>
Contact: <13>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"
04-08-2013 07:40 AM
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"
04-08-2013 07:53 AM
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.
04-08-2013 07:56 AM
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"
04-08-2013 08:10 AM
RTP event from actually working 7940:
1w6d: s=VoIP d=DSP payload 0x65 ssrc 0xE155D728 sequence 0x33E1 timestamp 0x1 FAC70
1w6d: <<
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: <<
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.
04-08-2013 08:19 AM
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=433A9D98-20C226805813>
To: <800123456>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>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-2722461213>
To: <800123456>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"192.168.105.21:5060>
Cisco-Guid: 2902615296-0000065536-0000000460-0359246016
Session-Expires: 1800
P-Asserted-Identity: "Marek JIRICEK" <13>13>
Remote-Party-ID: "Marek JIRICEK" <13>;party=calling;screen=yes;privacy=off13>
Contact: <13>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"
04-08-2013 08:31 AM
04-08-2013 08:43 AM
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"
04-09-2013 01:01 AM
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
04-09-2013 01:04 AM
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"
04-08-2013 07:50 AM
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
04-08-2013 07:59 AM
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.
04-08-2013 09:45 AM
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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