#1 show call active voice compact
Use case: We have an active call through the CUBE and we want to see the codec that is being used along with the media ip addresses and RTP port numbers.
R2# show call act voice comp A/O FAX T Codec type Peer Address IP R: Total call-legs: 2 307 ANS T13 ilbc VOIP P2001 192.102.64.50:24818 308 ORG T13 ilbc VOIP P3001 192.102.65.50:31562
#2 show call active video compact
Use case: We have an active VIDEO call through the CUBE and we want to see the codec that is being used, media ip addresses and RTP port numbers.
R1#sh call act vid comp A/O FAX T Codec type Peer Address IP R: Total call-legs: 2 13 ANS T14 H264 VOIP-VIDEO P82022002 192.102.64.30:20310 14 ORG T14 H264 VOIP-VIDEO P85151111 192.202.64.1:16988
#3 debug voip dialpeer
Use case: We have an H323/SIP PSTN gateway or CUBE and the call we are teting fails. We want to see which inbound and outbound dial-peer is being matched within IOS.
R2#deb voip dialp voip dialpeer default debugging is on ---------->8 Snippet 8< -------------- Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=972 - ---------->8 Snippet 8< -------------- List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=911
#4 show call active voice brief or show call active video brief
Use case: We have an active call through an IOS gateway/CUBE and would like to check which dial-peers have been matched.
R2#sh call act voice brief | i pid : ms. () + pid: 386E : 9 21766850ms.1 (17:51:25.988 cst Thu Mar 17 2016) +-1 pid:972 Answer 3001 connected 386E : 10 21766900ms.1 (17:51:26.036 cst Thu Mar 17 2016) +-1 pid:911 Originate 911 connecting
#5 debug ccsip messages
Use case: We have a SIP-based CUBE environment or any other kind of SIP gateway for that matter, we have checked that the correct dial-peers are being matched but the call still fails. We want to see the SIP messaging.
R1#deb ccsip mess SIP Call messages tracing is enabled R1# Aug 11 20:30:22.303: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:85151111@192.102.64.254:5060 SIP/2.0 Via: SIP/2.0/TCP 192.100.64.12:5060;branch=z9hG4bKc4c1028de0d From: "hq p2" <sip:82022002@192.100.64.12>;tag=8080~2c6d7370-f99d-4063-a09c-0c653a418ded-41540807 To: <sip:85151111@192.102.64.254>
#6 show sccp connection
Use case:: We are calling an application which does not support the low bit rate codec (g729 in the example below) and we want to verify that the transcoder is being invoked. Also can be used to verify conference bridge usage and also IOS MTP usage.
R3#sh sccp conn sess_id conn_id stype mode codec sport rport ripaddr conn_id_tx 65539 12 xcode sendrecv g729 16546 2000 192.102.66.254 65539 16 xcode sendrecv g711u 16544 2000 192.102.66.254 Total number of active session(s) 1, and connection(s) 2
#7 debug tftp events
Use case: We have are trying to register a SCCP or SIP phone to the CME but registration is failing. We can try and find out if the phone is initiating a TFTP GET in order to grab it’s xml configuration file and find out if the phone is able to download it.
R3#debug tftp events —————————–>8 Snippet 8< ————————— Jul 9 16:31:01.442: TFTP: Looking for SEP6886A7C43389.cnf.xml Jul 9 16:31:01.442: TFTP: Opened flash:/its/SEP6886A7C43389.cnf.xml, fd 4, size 4646 for process 392 Jul 9 16:31:01.454: TFTP: Finished flash:/its/SEP6886A7C43389.cnf.xml, time 00:00:00 for process 392 Jul 9 16:31:06.430:
#8 debug isdn q931
Use case: We are calling the PSTN over a ISDN Q931 PRI and we want to verify the .
R2#deb isdn q931 debug isdn q931 is ON. Mar 17 19:34:56.175: ISDN Se0/2/0:23 Q931: TX -> SETUP pd = 8 callref = 0x0082 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Display i = 'sb p1' Calling Party Number i = 0x0081, '9723033001' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '911' Plan:Unknown, Type:Unknown
#9 test voice translation-rule x
Use case: We have created a voice translation-profile to modify the Calling and Called Number and would like to verify if the syntax and logic of the voice translation-rules is correct.
R2#test voice translation-rule 1 9723033001 Matched with rule 1 Original number: 9723033001 Translated number: 9723033001 Original number type: none Translated number type: none Original number plan: none Translated number plan: none
#10 debug voice translation
Use case: We have a call through any IOS gateway/CUBE and we want to check if our voice translation-profiles are being matched.
R2#deb voice translation VoIP Translation Rule debugging is enabled R2#le_translate_internal: number=3001 type=unknown plan=unknown numbertype=calling Mar 17 23:44:49.722: //-1/00345A3B0600/RXRULE/regxrule_profile_match_internal: Matched with rule 1 in ruleset 9110 Mar 17 23:44:49.722: //-1/00345A3B0600/RXRULE/regxrule_profile_match_internal: Matched with rule 1 in ruleset 9110 Mar 17 23:44:49.722: //-1/00345A3B0600/RXRULE/sed_subst: Successful substitution; pattern=3001 matchPattern=^3...$ replacePattern=972303\0 replaced pattern=9723033001 Mar 17 23:44:49.722: //-1/0034 R2#5A3B0600/RXRULE/regxrule_subst_num_type: Match Type = any, Replace Type = unknown Input Type = unknown Mar 17 23:44:49.722: //-1/00345A3B0600/RXRULE/regxrule_subst_num_plan: Match Plan = any, Replace Plan = unknown Input Plan = unknown Mar 17 23:44:49.722: //-1/00345A3B0600/RXRULE/regxrule_profile_translate_internal: xlt_number=9723033001 xlt_type=unknown xlt_plan=unknown Mar 17 23:44:49.722: //-
Good stuff Vik. I use most of these every day!
very useful Vik.. thanks lot ..
One of my favorites that shows the calling and called party along with the matched dial-peer in real time is
#debug translation detail
Vik you are 1 of the most recognized person in industry.Thank you for posting these please share some more tips
Few other useful commands
show voip rtp connections
show sip-ua calls brief