Archiv der Kategorie: Unified Communications

SIP debugging overview

SIP debugging overview

debug ccsip: This has various options,

  • debug ccsip all:  This command enables all ccsip type debugging. This debug command is very active, you should use it sparingly in a live network
  • debug ccsip calls: This command displays all SIP call details as they are updated in the SIP call control block. You can use this debug command to monitor call records for suspicious clearing causes.
  • debug ccsip errors: This command traces all errors that are encountered by the SIP subsystem.
  • debug ccsip events: this command traces event, such as call setups, connections and disconnections. An events version of a debug command is often the best place to start because detailed debugs provide much useful information.
  • debug ccsip info: This command enables tracing of general SIP security parameter index (SPI) information, including verification that call redirection is disabled.
  • debug ccsip media: This command enables tracing of SIP media streams
  • debug ccsip messages: This command shows the headers of SIP messages that are exchanged between a client and a server.
  • debug ccsip preauth: This command enables diagnostic reporting of authentication, authorization, accounting (AAA) for SIP calls.
  • debug ccsip states: This command displays the SIP states and state changes for sessions within the SIP subsytem.
  • debug ccsip transport: This command enables tracing the SIP transport handler and the TCP or UDP process

debug voip ccapi inout: This command shows every interaction with the call control application programming interface (API) on both the telephone interface and on the VOIP side. By monitoring the output, you can follow the progress of a call from the inbound interface or VOIP peer to the outbound side of the call. This debug command is very active. you should use it sparingly in a live network.

debug voip ccpai protoheaders: This command displays messages that are sent between the originating and terminating gateways. If no headers are being received by the terminating gateway, verify that the header-passing command is enabled on the originating gateway.

Feature Design of SIP Debug Output Filtering Support

Prior to the SIP Debug Output Filtering Support feature, debugging and troubleshooting on the VoIP gateway was made more challenging by the extensive amounts of raw data generated by debug output.

This feature allows the debug output for a SIP call to be filtered according to a variety of criteria. The SIP Debug Output Filtering Support feature provides a generic call filtering mechanism that does the following:

•Allows you to define a set of matching conditions for filtering calls.

•Identifies the desired calls based on the configured matching conditions inside VoIP gateways.

•Coordinates the filtering effort on traced calls between multiple modules inside VoIP gateways.

•Displays the debugging trace for calls that match the specified conditions.

SIP Debug Commands that Support Output Filtering

•debug ccsip all

•debug ccsip calls

•debug ccsip events

•debug ccsip messages

•debug ccsip preauth

•debug ccsip states

Configuring Call Filters

This task configures the conditions for filtering SIP calls.


1. enable

2. configure terminal

3. call filter match-list number voice

4. incoming calling-number string

5. incoming called-number string

6. incoming signaling {local | remote} ipv4 ip-address

7. incoming media {local | remote} ipv4 ip-address

8. incoming dialpeer tag

9. outgoing calling-number string

10. outgoing called-number string

11. outgoing signaling {local | remote} ipv4 ip-address

12. outgoing media {local | remote} ipv4 ip-address

13. outgoing dialpeer tag

14. end


call filter match-list 1 voice
incoming called-number 4085559876
voice-port 0:D
voice-port 1:D
voice-port 2:D
voice-port 3:D

Enabling SIP Debug Output Filtering: Example

Router# debug condition match-list 1 exact-match
Router# debug ccsip all

Router# show debug

CCSIP SPI:SIP Call Statistics tracing is enabled       (filter is ON)
CCSIP SPI:SIP Call Message tracing is enabled  (filter is ON)
CCSIP SPI:SIP Call State Machine tracing is enabled    (filter is ON)
CCSIP SPI:SIP Call Events tracing is enabled   (filter is ON)
CCSIP SPI:SIP error debug tracing is enabled   (filter is ON)
CCSIP SPI:SIP info debug tracing is enabled    (filter is ON)
CCSIP SPI:SIP media debug tracing is enabled   (filter is ON)
CCSIP SPI:SIP Call preauth tracing is enabled  (filter is ON)

Router# Debug filtering is now on
Building configuration…
call filter match-list 1 voice
incoming called-number 4085551221

Cisco Unity Connection Audio File Format

Kennen Sie das richtige Audio File Format für den Upload in Cisco Unity Connection ?
Wenn nicht, dann werden die folgenden Angaben weiterhelfen um das richtige File Format zu bestimmen und auch mit welcher Software Sie eine Konvertierung vornehmen können.

Das richtige File Format welches von Cisco Unity Connection akzeptiert wird ist:

Audio Format: CCITT u-Law
Bit Rate: 64kbps
Sample Size: 8 bit
Channels: 1 (mono)
Sample Rate:  8 khz

Wenn Ihre Aufnahme nicht im richtigen Format vorliegt ist es einfach, diese zu konvertieren.
Nachfolgend liste ich die Schritte, beispielhaft für die Software Wave Pad auf.

  • Download der Software von der offiziellen Wavepad WebSite
  • Installiere die Software auf einem Windows PC
  • Starte die Software
  • Klick File > Open File > suchen des zu konvertierenden Audio Files > Open
  • Klick File > Save File As > Filenamen eingeben > Save
  • Im nachfolgendem Fenster die Wave Encoder Options wie folgt abändern
  • Settings > Custom
  • Encoding > CCITT u-Law
  • Format > 8.000 KHz, 8 Bit, Mono
  • klick OK

How to add DHCP Scope Option 150 for Cisco CallManager

By default, the DHCP Scope option 150 for Cisco TFTP Server is not available on Windows Server. Below are the steps to add the configure the option 150:

1.  First we need to define the option 150 so that it can be made available in the scope options list. For this, right click on the IPv4and choose the option  Set Predefined Options… (screenshot below)


2.  Click on the Add… button and you will get another dialog box asking you the Option Type.

3.  Fill the details as mentioned below and shown in the screenshot:

Name: Cisco TFTP Server
Data Type: IP Address
Array: Checked
Code: 150
Description: Used for Cisco Call Manager TFTP Server 


4.  Click OK and click Edit Array button to enter the IP address for the TFTP Server (Screenshot below). Once you are done, click OK to exit.


5.  Now, to configure Option 150 for any scope, go to its Scope Options, right click on and choose Configure Options and select the Option 150 from the list. You can edit the IPs if you required.


Login Unavailable (23) error message with extension mobility

(STANDARD extension mobility, not cross-cluster)

Eventuell habt ihr bei Eurer extension mobility“ Konfiguration im Cisco unified communication manager auch schon den Error beim einloggen auf einem Telefon erhalten:

Login Unavailable (23)

Bei der Fehlersuche im Netz findet man sehr viele Einträge zum cross-cluster Feature, aber hier geht es mir um Standard extension mobility.

Das Problem lässt sich dadurch lösen, dass man in der End User Konfiguration einen Haken an „Home Cluster“ setzt.

User Management -> End User, click Home Cluster checkbox