Asterisk ztdummy compiled, but not loaded, even afer asteri

I compiled and installed the Asterisk ztdummy package because there is no rpm for it, unfortunately, and i even reinstalled asterisk, but i still get the “No application ‘Meetme’ for extension…” error when trying to conference. I do a “module show”, and it lists other modules that were compiled with the zt source, but not ztdummy.

Does anyone know how to fix this? This is more than a passing interest or hobby, because i need to conference about 3 to 5 people to help me test a new Website Content Management System and User Forums Management System i am about to launch as a service.

Fedora 10 x86_64


    -- Unregistered SIP 'Vector'
    -- Registered SIP 'Vector' at xx.xx.xx.xx port 5061
       > Saved useragent "Twinkle/1.4.2" for peer Vector
[Sep 13 01:23:05] NOTICE[29436]: chan_sip.c:15373 handle_response_peerpoke: Peer 'Vector' is now Reachable. (4ms / 2000ms)
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Executing [xxx@office:1] Answer("SIP/Vector-0197ca10", "") in new stack
    -- Executing [xxx@office:2] Wait("SIP/Vector-0197ca10", "1") in new stack
    -- Executing [xxx@office:3] Authenticate("SIP/Vector-0197ca10", "xxx") in new stack
    -- <SIP/Vector-0197ca10> Playing 'agent-pass.ulaw' (language 'en')
    -- <SIP/Vector-0197ca10> Playing 'auth-thankyou.ulaw' (language 'en')
[Sep 13 01:23:13] WARNING[29458]: pbx.c:3082 pbx_extension_helper: No application 'MeetMe' for extension (office, xxx, 4)
  == Spawn extension (office, xxx, 4) exited non-zero on 'SIP/Vector-0197ca10'

and i have “load =>” in modules.conf

Hate*CLI> module show
Module                         Description                              Use Count                String handling dialplan functions       0                    Simplified Message Desk Interface (SMDI) 0               HTTP Phone Provisioning                  0                  Generic Speech Recognition API           0             Region-specific tones                    0               share-able code for AEL                  0                 Call Monitoring Resource                 0                     Asterisk Gateway Interface (AGI)         1             Music On Hold Resource                   0                     Simple FAX Application                   0              Dialgroup dialplan function              0                     Database (astdb) related dialplan functi 0                     SMS/PSTN handler                         0               Dialplan Context/Extension/Priority Chec 0                    ADSI Resource                            0                 Raw H.263 data                           0                Realtime Data Lookup/Rewrite             0                   Random number dialplan function          0           OGG/Vorbis audio                         0                Stores output of file into a variable    0                Indicator for whether a voice mailbox ha 0                     Send URL Applications                    0               Read/Write/Store/Destroy values from a R 0                  Cryptographic Digital Signatures         0                   Extension Macros                         0              Hangs up the requested channel           0                    Read Variable Application                0                    URI encode/decode dialplan functions     0              Microsoft WAV format (Proprietary GSM)   0                Realtime Switch                          0               Digital Milliwatt (mu-law) Test Applicat 0                   GSM Coder/Decoder                        0                 Listen to the audio of an active channel 0                Asterisk ADSI Programming Application    0                   True Call Queueing                       0                 Bluetooth Mobile Device Channel Driver   0                   Dialplan mutexes                         0         Redirects a given channel to a dialplan  0                  ITU G.722-64kbps G722 Transcoder         0          Wait For Silence                         0                  Local Proxy Channel (Note: used internal 0                 Virtual Dictation Machine                0                    Interface Test Application               0                 Technology independent volume control    0             External IVR Interface Application       0                  Generic System() application             0                 Raw G729 data                            0                 JPEG (Joint Picture Experts Group) Image 0                 Asterisk Manager Interface CDR Backend   0               Feature Proxy Channel                    0               Read and evaluate extension validity     0             Waits until first ring after time        0                  Dialogic VOX (ADPCM) File Format         0                Send DTMF digits Application             0            Dialplan Speech Applications             0             Say time                                 0                    Call Detail Record (CDR) dialplan functi 0                Send Text Applications                   0              Channel Pickup Application               0             Check channel availability               0                    Environment/filesystem dialplan function 0                Transfers a caller to another extension  0                Sound File Playback Application          0                 Variable dialplan functions              0                   SHA-1 computation dialplan function      0               Mixed Audio Monitoring Application       0                  Silly NBS Stream Application             0                  Charset conversions                      0                Loopback Switch                          0                  A-law and Mulaw direct Coder/Decoder     0                 Raw G.726 (16/24/32/40kbps) data         0                   Outgoing Spool Support                   0                    Dialing Application                      0                   While Loops and Conditional Execution    0                    Simple Echo Application                  0                 base64 encode/decode dialplan functions  0                  Microsoft WAV format (8000Hz Signed Line 0                    Executes dialplan applications           0                  Raw Signed Linear Audio support (SLN)    0                Dump Info About The Calling Channel      0                 Checks if Asterisk module is loaded in m 0                     Asterisk Extension Language Compiler     0                 Send verbose output                      0                   Resource limits                          0              SLIN Resampling Codec                    0              Block Telemarketers with Special Informa 0             Channel group dialplan functions         0         Directed Call Pickup Application         0                  Text Extension Configuration             0                  Customizable Comma Separated Values CDR  0                  A-law Coder/Decoder                      0                Channel information dialplan function    0                    Cut out information from a string        0             Set CallerID Presentation Application    0                    DISA (Direct Inward System Access) Appli 0                   Distributed Universal Number Discovery ( 0           Alarm Receiver for Asterisk              0                  Logical dialplan functions               0                  Raw GSM data                             0                 Raw H.264 data                           0                      Database Access Functions                0                 File format conversion CLI command       0                   Image Transmission Application           0                  ITU G.726-32kbps G726 Transcoder         0                  mu-Law Coder/Decoder                     0            Authentication Application               0                    Session Initiation Protocol (SIP)        0         Comedian Mail (Voicemail System)         0         Extension Directory                      0                 G.723.1 Simple Timestamp File Format     0                   Inter Asterisk eXchange (Ver 2)          0                  Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0               Caller ID related dialplan functions     0                Channel timeout dialplan functions       0                 Require phone number to be entered, if n 0                Find-Me/Follow-Me Application            0                Get ADSI CPE ID                          0               Morse code                               0               Wait until specified time                0                 Speex Coder/Decoder                      0               Custom User Event Application            0                     Comma Separated Values CDR Backend       0                 Adaptive Differential PCM Coder/Decoder  0                   Mathematical dialplan function           0                System information related functions     0            Call origination from the CLI            0                  Trivial Record Application               0              Look up Caller*ID name/number from black 0                   Media Gateway Control Protocol (MGCP)    0                   Dialplan subroutines (Gosub, Return, etc 0               Gets an extension's state in the dialpla 0                    OSS Console Channel Driver               0                   ENUM related dialplan functions          0               Gets or sets a device state in the dialp 0                 Fork The CDR into 2 separate entities    0                Raw Signed Linear 16KHz Audio support (S 0                  Returns the output of a shell command    0                  Linux Telephony API Support              0         Call Parking and Announce Application    0                Simple Festival Interface                0         Control Playback Application             0                    MD5 digest dialplan functions            0                     Tell Asterisk to not maintain a CDR for  0                  Agent Proxy Channel                      0                     Answering Machine Detection Application  0                 LPC10 2.4kbps Coder/Decoder              0              Playback with Talk Detection             0                Get Asterisk Version/Build Info          0         
148 modules loaded

For a new installation, you should be using dahdi-dummy.

My guess is that you haven’t started the device driver.

ok, so where can i get dahdi-dummy from? and what do you mean that i haven’t started it? It is a module, nothing more, right? Because i don’t have the hardware, i only use SIP softphones. I’m new, so i need more details.


dahdi dummy is part of dahdi, which is the replacement for zaptel in all current versions of Asterisk.

zaptel/dahdi is a device driver and needs to be loaded into the kernel before Asterisk is started. It is a kernel module, not an Asterisk one. On CentOS, for dahdi, it is started by /etc/rc.d/init.d/dahdi, which the proper install process will arrange to be automatically run on boot up.

i did a “yum list dahdi” after i seen your post, and found some packages, and did a “yum -y installed asteriskdahdi dahdi-tools” and seen that it also installed zaptel-libs, so i think this may fix the problem, i will update this topic in a few minutes…


If you could use yum, you are not using plain Asterisk!

Oh, ok, well i started the dahdi service and restarted asterisk, and now i get further along :smiley:. But i’m still not sure that i get far enough. Here is the console output that resulted from dialing the conference extension (999):

-- <SIP/Vector-00c01a90> Playing 'agent-pass.ulaw' (language 'en') -- <SIP/Vector-00c01a90> Playing 'auth-thankyou.ulaw' (language 'en') -- Executing [999@office:4] MeetMe("SIP/Vector-00c01a90", "18|pM") in new stack == Parsing '/etc/asterisk/meetme.conf': == Found == Spawn extension (office, 999, 4) exited non-zero on 'SIP/Vector-00c01a90' Hate*CLI>

and here is my meetme.conf (this password is completely ignored for some reason, and it uses the password in the dial plan…)

[rooms] conf => 999,666,123456789

in /etc/dahdi/system.conf everything is commented out except for “loadzone = us” and “defaultzone=us”.

In /etc/dahdi/modules here are the modules that are not commented out wct4xxp wcte12xp wct1xxp wcte11xp wctdm24xxp wcfxo wctdm xpp_usb

I just wonder if there is something else that i’m missing…

thanx again!

what do you mean with your previous post? I’m using the latest asterisk on Fedora 10 (very similar to centOS/Red Hat).

The CLI trace doesn’t show any errors.

So how do i get to wait around in the conference room for people to dial in, because after i enter the password, it says “thank you”, and then hangs up? Here is my dial plan for the conference room:

exten => 999,1,Answer
exten => 999,n,Wait(1)
exten => 999,n,Authenticate(xxx)
exten => 999,n,MeetMe(18|pM)
exten => 999,n,Playback(vm-goodbye)
exten => 999,n,Hangup