Debugging inbound call - need debug command

I am debugging what I believe to be a dial plan issue on PGW but want to run a debug command on the ITP to show the actual called number of the inbound call.
What is the best debug command to use to see the inbound called and calling numbers of a call.

You are better off capturing an mdl trace from the PGW using per call tracing or using snoop and opening the call in Wireshark.
The ITP will only dump hex values and you will need to decode them yourself since it is not concerned with ISUP layer. You can use the cs7 paklog feature to attach an access list to a linkset and then send the raw data to a syslog server for decoding.
For example:
cs7 paklog x.x.x.x dest-port xxxx src-port yyyy
access-list 2700 instance x permit si all
debug cs7 mtp3 paklog 2700

Similar Messages

Debug command for FXO digits on called number

I'd like to be able to debug the digits being passed out an FXO port during an external call.
Does anyone know the best debug command to use?
thanks in advance for the help.

Hi friend,
You can use the debug voip ccapi inout, check this link, may be helpful:
http://www.cisco.com/en/US/docs/ios/12_3/debug/command/reference/dbg_v1g.html#wp1106585
Regards,
-adriÃ¡n.

Gateway crashed after debug command - how to recover

Hi - I was troubleshooting an issue out of hours and the voice gateway I was working on crashed after using a debug command
Whats the best way to recover the gateway?
I'm unable to ping it now - do I need someone to reboot it?
Once it comes back up with debugging still be active ?

Chris if the debug you entered is too processor intensive then the only way to recover is to reboot. After Reboot the debugs are turned off.
Sometimes it becomes too slow but you are still able to access it intermittently if thats the case then you can wait for it become accessible for a small period of time and try to quikly enter u all (undebug all) otherwise if its completely inaccesible your only way to recover is to reboot.
-Terry

Debug commands to troubleshoot client through the WiSM

We been having issues with several laptops that appear to be losing wireless connectivity and i wanted to know if there is a debug command or commands that would allow me to see the process of the client trying to Associate/Authenticate, etc..
Thank you
Vic

This may be a separate question, I am quite familiar with the "debug client" command, however everytime I log out, or am logged out of the console, and then log back in, my "show debug", shows that I am not debugging anything. The client mac is listed, but no Debug Flags are enabled. Is there no way to leave those debug commands on, so the output is put out to a syslog server? Thanks

[Debugging] - how to avoi intensive CPU usage with debug command

Hi all !
I hope you're well !
I'm just wondering if there is a mean to avoid the increase of CPU usage when using the debug command. More precisely I would like to know if it exists something to avoid the CPU to saturate and to allow vital process to be prioritary for using the CPU, just like a QoS mecanism but to classify the process in order of importance.
Thanks a lot in advance !

some time ago this document was posted on CSC which give an excellent overview on how to savely enable debug commands:
https://supportforums.cisco.com/docs/DOC-16310

How to use CTC debug command?

As be known that hold ctrl+shift while clicks "help -> about CTC" above the CTC window resulting the debug window occurs.
Does anyone know the debug command usage?
Since we got a provisioning issue on the 15454-CE card which exists a "hidden" crossconnect on a port, and not shown on CTC.
Any advice will be appreciated!
Ricky

The best way to look for cross connects is via TL1. Open the TL1 window and log in:
act-user:::$::;
rtrv-crs:::$;
The hard part is getting the AID correct. For example, the old style OC48 cards would be "STS-6-1-all". You will have to spend some time in the TL1 manual to get the correct AID.
Hope this helps,
Tim

Debug commands for CatOs

I am looking for some debug commands for CatOs for my 4006 or 6509 running CatOs.
please!

Perhaps you could tell us what the problem is and we can lead you in the right direction. 'debug' as known in IOS isn't really there in CatOS.

Debug command adsl traffic

cisco 2651XM router
IOS: c2600-ipvoicek9-mz.124-15.T7.bin
wic1-adsl card fitted
which debug command will give me detailed info about traffic through the adsl connection? Specifically I want to see originating ip and destination ip of internet traffic (much like a wireshark readout)
I tried debug atm events and debug atm packet but these didn't contain the info I was after. Thanks for any pointers.

my router config doesn't specify anywhere which PPPo I'm using. All it says under Dialer0 is encapsulation ppp. I did the terminal from the console port using privlelged exec but it displays nothing when using 'debug ppp packet'.
ps: hold on, if I do
#show int dialer0
Dialer0 is up, line protocol is up (spoofing)
10345847 packets input, 229783612 bytes
7537931 packets output, 2120477085 bytes
Bound to:
Virtual-Access2 is up, line protocol is up
PPPoATM vaccess, cloned from Dialer0
does this mean I'm using PPPoA?

Inbound Calls Fail

HI.
I'm running CME on a 2851.
My issue is with inbound calls not failing totally, but then requiring the extension to be dialed. This only happens for calls where the calling number is unknown.
So all calls inbound work if the inbound calling number is known. Calls route nicely.
If the calling number is unknown the caller gets a tone, whereby they need to then dial the extension.
Ideally I'd like to have unknown calling numbers simply route the call the same way as with a known calling number.
FYI when i debug q931 caller id (if off) is shown as Calling Party Number i = 0x00A3, N/A
Appreciate any help as this is causing some problems!!!
Configuration attached

Timothy,
When the calls come in, a dial-peer match is made, and in this case, where the calling number is known, the dial-peer whose destination-pattern command matches the calling number is selected, and the call is routed.
For calls where the calling number is not available, there is no dial-peer matched, so the default dial-peer (dial-peer 0) is used, and two stage dialing is then required.
To resolve, a quick solution would be to create a new dial-peer that has the incoming called-number command (which matches on the called number), so that this dial-peer will always be used for all incoming calls regardless of whether the calling number is available or not.
Use the direct-inward-dial command as well in this dial-peer to force one stage dialing.
dial-peer voice 90 pots
incoming called-number .
direct-inward-dial
port 0/0/0:15.
See also
* http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#topic3
* http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800e00d0.shtml
Hope this helps.
Regards,
Michael.

Inbound calls getting answered automatically in Cisco CME

Hi All....
Please advise me the reason, why the inbound calls to CME got answered automatically.?
I am mentioning the call handling scenario here with..
Trunk Type : FXO
Connection PLAR to physical extension(200)
timeouts call-disconnect 1
timeouts wait-release 1
voice-port 0/0/0
trunk-group FXO
cptone AE
timeouts call-disconnect 1
timeouts wait-release 1
connection plar 200
caller-id enable
More over I tried with the below configuration also!!!!!!!!!!!! Bad luck !!
voice class custom-cptone UAE-TONE
dualtone disconnect
frequency 400
cadence 400 350 225 525
voice-port 0/1/0
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-TONE
timeouts call-disconnect 1
timeouts wait-release 1
dial-peer voice 202 pots
incoming called-number .
direct-inward-dial

The command connection plar 200 under voice port 0/0/0 directs the H323 gateway to automatically answer the inbound call on that FXO port and to attempt to transfer the call to DN 200.
If that is not what you want, then you need to remove the connection plar command and replace it with the command "secondary dialtone"

Inbound calls on AS5400

Hi,
I have a AS5400HPX device with a AS5400 T1 2 PRI DFC card installed and T1 link. I need to check the inbound calls on the T1 PRI link.
1) What commands can help me check verify if the call lands on the router, and what is the source number which is dialling?
2) Do all the Async interfaces on the device represent each of the client dailling in? How do I check which Async interface has been used by the client?
3) I will be using a local pool for assigning IP addresses to each of the client dialling in.Would this command suffice?
ip local pool <pool name> IP range
4) Also, what are these resource-pool group and resource-pool profile commands used for?
Appreciate your inputs.
Thanks
Mikey

Appreciate if someone could reply to this.
Cheers
Mikey

Configure SPA8800 for inbound calls in switzerland

Good Evening,
i have a problem with this gateway, i'm able to make correctly outbound calls but if i receive an inbound call from the pstn line it doesn't answer.. I saw that the psnt status reamains on "idle".
I'm in Swizterland (Lugano). How can i manage that?
Regards

Wrong line setting may cause ring is not recognized. Unfortunatelly, I know no correct setting for your Telco operator. Wait for someone's reply here or ask your operator for PSTN line specification. Also, seller of SPA8800 should support you. The European Telecomunication Standard ETS 300 001, table 1.7.9 (and others) may help you as well.
Now something completelly different. Are you sure you have call routing set properly ? If incomming calls have nowhere to go, then it will not be answered. Turn on syslog, debug log and catch them. It may reveal you something helpfull.

SIP Trunk not accepting inbound calls

I have a CME setup using Engin as a SIP provider
I am able to dial out with no issue, however my inbound calls do not work, they divert to the Engin voicemail
My SIP registration is OK and the number is configured as the primary DN on one of my phones
Router#sh sip-ua register status
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
038682XXXX                       -1         1124         yes
101                              20001      45           no
102                              20003      18           no
103                              20005      45           no
104                              20006      45           no
I do see the call come in if I debug the dial peer, but it only seems to match an outgoing dp
I am seeing a couple of disconnect cause codes that I cant seem to find any relavent information on in the CCSIP debugs
Router#
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:4F947560
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:uas, Min-SE Value:1800, flags:2001
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:4F947DF8
Sep 8 18:15:15.025: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:403, container:4F947B38
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4C39C570
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0417XXXXXX
Called Number            : 038682XXXX
Source IP Address (Sig ): 211.30.48.136
Destn SIP Req Addr:Port : 203.161.164.69:5060
Destn SIP Resp Addr:Port : 203.161.164.69:5060
Destination Name         : 203.161.164.69
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 211.30.48.136
Source IP Port    (Media): 17768
Destn IP Address (Media): 203.161.164.69
Destn IP Port    (Media): 18314
Orig Destn IP Address:Port (Media): [ - ]:0
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 21
Disconnect Cause (SIP)   : 403
Any Ideas
Doug

Hi Tapan,
Firstly the topology is as follows
ISP/VOIP provider - Internet - Cable modem - 2800 CME router - IP Phone
The VM is provided by the ISP
debug ccsip messages
Sep 9 10:21:41.488: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Contact:
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: multipart/mixed,application/media_control+xml,application/sdp
Min-SE: 60
Session-Expires: 1800;refresher=uas
Max-Forwards: 9
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 317
v=0
o=BroadWorks 18275729 1 IN IP4 203.161.164.69
s=-
c=IN IP4 203.161.164.69
t=0 0
m=audio 18128 RTP/AVP 18 8 0 101
c=IN IP4 203.161.164.69
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=bsoft: 1 image udptl t38
Sep 9 10:21:41.508: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Sep 9 10:21:41.516: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=21
Content-Length: 0
Sep 9 10:21:41.544: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
CSeq: 633854439 ACK
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
Max-Forwards: 9
Content-Length: 0
Voice Config
Router#
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server expires max 3600 min 3600
localhost dns:mel.byo.engin.com.au
no call service stop
voice class codec 1
codec preference 1 g711ulaw
voice translation-rule 10
rule 1 /^0/ //
voice translation-rule 11
rule 1 /^.*/ /0386821234/
voice translation-profile PSTN_Outgoing
translate calling 11
voice-card 0
dsp services dspfarm
mgcp profile default
sccp local Vlan100
sccp ccm 10.1.100.1 identifier 1 version 7.0
sccp
sccp ccm group 1
bind interface Vlan100
associate ccm 1 priority 1
associate profile 1 register confdsp
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
dial-peer voice 99 voip
translation-profile outgoing PSTN_Outgoing
destination-pattern .T
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 100 voip
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number 0386821234
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 110 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number .%
dtmf-relay rtp-nte
no vad
dial-peer voice 90 voip
description Melbourne 03 Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [89].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 91 voip
description National Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 0[278]........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 92 voip
description Vic/Tas 03 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [56].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 93 voip
description Mobile numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 04........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 94 voip
description 13XXXX numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 13[1-9]...
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 96 voip
description 1300/1800 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 1[38]00......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 98 voip
description Emergency 000
translation-profile outgoing PSTN_Outgoing
destination-pattern 000
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
sip-ua
credentials username 0386821234 password 7 XXXX realm voice.mibroadband.com.au
authentication username 0386821234 password 7 XXXX
nat symmetric role active
nat symmetric check-media-src
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:mel.byo.engin.com.au expires 3600 port 5060 transport udp unsolicited
registrar dns:mel.byo.engin.com.au expires 3600
sip-server dns:mel.byo.engin.com.au
connection-reuse
telephony-service
sdspfarm conference mute-on #1 mute-off #2
sdspfarm units 2
sdspfarm tag 1 confdsp
conference hardware
max-ephones 42
max-dn 144
ip source-address 10.1.100.1 port 2000
calling-number initiator
service phone videoCapability 1
service phone displayOnDuration 00:01
service phone displayOnTime 08:30
service phone displayOffTime 17:30
service phone displayIdleTimeout 00:01
service phone displayOnWhenIncomingCall 1
system message Cisco CME
load 7941 SCCP41.8-4-2S
load 7942 SCCP42.8-4-2S
load 7945 SCCP45.8-4-2S
load 7961 SCCP41.8-4-2S
load 7962 SCCP42.8-4-2S
load 7965 SCCP45.8-4-2S
load ata ATA030204SCCP090202A
time-zone 48
date-format dd-mm-yy
voicemail 90125200
mwi relay
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh music-on-hold.au
web admin system name cisco secret 5 $1$d8/H$glhLiCCWXmFSUp6BtwGho0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 0.T
create cnf-files version-stamp 7960 Jul 06 2011 10:32:45
ephone-dn 1 dual-line
number 038682XXXX
label 101
name 7965
mwi sip
ephone-dn 2 dual-line
number 102
label 102
name 7941
ephone-dn 3 dual-line
number 103
label 103
name 7920
ephone 1
device-security-mode none
video
mac-address 0023.5EB8.6E4E
type 7965
button 1:2 2:1
ephone 3
device-security-mode none
mac-address 0019.0633.A933
max-calls-per-button 2
type 7920
button 1:3
ephone 10
device-security-mode none
mac-address 0019.E7B7.BAB3
max-calls-per-button 2
type ata
button 1:1

The inbound calls to our call center is drop after putting it on hold or transfer

Dear All;
Good day
The inbound calls to our call center is drop after putting it on hold or transfer the call to another agent. The MOH file is playing till 21 sec only then call drop . the agent cant resume the call again. The MOH file is running from Gateway (multicast).
No problem in outbound calls.
I urgent need you help
Should you require any more information , please do not hesitate to contact me.
Thanks & Best Regards,
Muhammad Fathy,
IT Network Manager
ALEXBANK
A subsidiary of Intesa Sanpaolo Group
Head office: B210-F1, Smart Village , KM 28 Cairo-Alex Desert Road, Egypt.
Cell:      +201017288844.
Office: +202-35311300   Ext: 8090.
eMail:   [email protected]
i To maintain a paperless environment, please don't print this e-mail unless you really need to.

Typically you have a codec or media resource issue to track down. IE, MTP, region, location, gateway trunk to trunk to call or something in that area.   Bypass UCCX and do the same call without this app... does it happen with a normal call?

How to call Operating System commands / external programs from within APEX

Hi,
Can someone please suggest how to call Operating Systems commands / external programs from within APEX?
E.g. say I need to run a SQL script on a particular database. SQL script, database name, userid & password everything is available in a table in Oracle. I want to build a utility in APEX where by when I click a button APEX should run the following
c:\oracle\bin\sqlplusw.exe userud/password@database @script_name.sql
Any pointers will be greatly appreciated.
Thanks & Regards,

Hi Guys,
I have reviewed the option of using scheduler and javascript and they do satisfy my requirements PARTIALLY. Any calls to operating system commands through these features will be made on the server where APEX is installed.
However, here what I am looking at is to call operating systems programs on client machine. For example in my APEX application I have constructed the following strings of commands that needs to be run to execute a change request.
sqlplusw.exe user/password@database @script1.sql
sqlplusw.exe user/password@database @script2.sql
sqlplusw.exe user/password@database @script3.sql
sqlplusw.exe user/password@database @script4.sql
What I want is to have a button/link on the APEX screen along with these lines so that when I click that link/button this entire line of command gets executed in the same way it would get executed if I copy and paste this command in the command window of windows.
Believe me, if I am able to achieve what I intend to do, it is going to save a lot of our DBAs time and effort.
Any help will be greatly appreciated.
Thanks & Regards,

Debugging inbound call - need debug command

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