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

Thanx

Hate*CLI> 
    -- 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'
Hate*CLI> 

and i have “load => app_meetme.so” in modules.conf

Hate*CLI> module show
Module                         Description                              Use Count 
func_strings.so                String handling dialplan functions       0         
res_smdi.so                    Simplified Message Desk Interface (SMDI) 0         
res_phoneprov.so               HTTP Phone Provisioning                  0         
res_speech.so                  Generic Speech Recognition API           0         
res_indications.so             Region-specific tones                    0         
res_ael_share.so               share-able code for AEL                  0         
res_monitor.so                 Call Monitoring Resource                 0         
res_agi.so                     Asterisk Gateway Interface (AGI)         1         
res_musiconhold.so             Music On Hold Resource                   0         
app_fax.so                     Simple FAX Application                   0         
func_dialgroup.so              Dialgroup dialplan function              0         
func_db.so                     Database (astdb) related dialplan functi 0         
app_sms.so                     SMS/PSTN handler                         0         
func_dialplan.so               Dialplan Context/Extension/Priority Chec 0         
res_adsi.so                    ADSI Resource                            0         
format_h263.so                 Raw H.263 data                           0         
res_realtime.so                Realtime Data Lookup/Rewrite             0         
func_rand.so                   Random number dialplan function          0         
format_ogg_vorbis.so           OGG/Vorbis audio                         0         
app_readfile.so                Stores output of file into a variable    0         
func_vmcount.so                Indicator for whether a voice mailbox ha 0         
app_url.so                     Send URL Applications                    0         
func_realtime.so               Read/Write/Store/Destroy values from a R 0         
res_crypto.so                  Cryptographic Digital Signatures         0         
app_macro.so                   Extension Macros                         0         
app_softhangup.so              Hangs up the requested channel           0         
app_read.so                    Read Variable Application                0         
func_uri.so                    URI encode/decode dialplan functions     0         
format_wav_gsm.so              Microsoft WAV format (Proprietary GSM)   0         
pbx_realtime.so                Realtime Switch                          0         
app_milliwatt.so               Digital Milliwatt (mu-law) Test Applicat 0         
codec_gsm.so                   GSM Coder/Decoder                        0         
app_chanspy.so                 Listen to the audio of an active channel 0         
app_adsiprog.so                Asterisk ADSI Programming Application    0         
app_queue.so                   True Call Queueing                       0         
chan_mobile.so                 Bluetooth Mobile Device Channel Driver   0         
func_lock.so                   Dialplan mutexes                         0         
app_channelredirect.so         Redirects a given channel to a dialplan  0         
codec_g722.so                  ITU G.722-64kbps G722 Transcoder         0         
app_waitforsilence.so          Wait For Silence                         0         
chan_local.so                  Local Proxy Channel (Note: used internal 0         
app_dictate.so                 Virtual Dictation Machine                0         
app_test.so                    Interface Test Application               0         
func_volume.so                 Technology independent volume control    0         
app_externalivr.so             External IVR Interface Application       0         
app_system.so                  Generic System() application             0         
format_g729.so                 Raw G729 data                            0         
format_jpeg.so                 JPEG (Joint Picture Experts Group) Image 0         
cdr_manager.so                 Asterisk Manager Interface CDR Backend   0         
chan_features.so               Feature Proxy Channel                    0         
app_readexten.so               Read and evaluate extension validity     0         
app_waitforring.so             Waits until first ring after time        0         
format_vox.so                  Dialogic VOX (ADPCM) File Format         0         
app_senddtmf.so                Send DTMF digits Application             0         
app_speech_utils.so            Dialplan Speech Applications             0         
app_sayunixtime.so             Say time                                 0         
func_cdr.so                    Call Detail Record (CDR) dialplan functi 0         
app_sendtext.so                Send Text Applications                   0         
app_pickupchan.so              Channel Pickup Application               0         
app_chanisavail.so             Check channel availability               0         
func_env.so                    Environment/filesystem dialplan function 0         
app_transfer.so                Transfers a caller to another extension  0         
app_playback.so                Sound File Playback Application          0         
func_global.so                 Variable dialplan functions              0         
func_sha1.so                   SHA-1 computation dialplan function      0         

app_mixmonitor.so              Mixed Audio Monitoring Application       0         
app_nbscat.so                  Silly NBS Stream Application             0         
func_iconv.so                  Charset conversions                      0         
pbx_loopback.so                Loopback Switch                          0         
codec_a_mu.so                  A-law and Mulaw direct Coder/Decoder     0         
format_g726.so                 Raw G.726 (16/24/32/40kbps) data         0         
pbx_spool.so                   Outgoing Spool Support                   0         
app_dial.so                    Dialing Application                      0         
app_while.so                   While Loops and Conditional Execution    0         
app_echo.so                    Simple Echo Application                  0         
func_base64.so                 base64 encode/decode dialplan functions  0         
format_wav.so                  Microsoft WAV format (8000Hz Signed Line 0         
app_exec.so                    Executes dialplan applications           0         
format_sln.so                  Raw Signed Linear Audio support (SLN)    0         
app_dumpchan.so                Dump Info About The Calling Channel      0         
func_module.so                 Checks if Asterisk module is loaded in m 0         
pbx_ael.so                     Asterisk Extension Language Compiler     0         
app_verbose.so                 Send verbose output                      0         
res_limit.so                   Resource limits                          0         
codec_resample.so              SLIN Resampling Codec                    0         
app_zapateller.so              Block Telemarketers with Special Informa 0         
func_groupcount.so             Channel group dialplan functions         0         
app_directed_pickup.so         Directed Call Pickup Application         0         
pbx_config.so                  Text Extension Configuration             0         
cdr_custom.so                  Customizable Comma Separated Values CDR  0         
codec_alaw.so                  A-law Coder/Decoder                      0         
func_channel.so                Channel information dialplan function    0         
func_cut.so                    Cut out information from a string        0         
app_setcallerid.so             Set CallerID Presentation Application    0         
app_disa.so                    DISA (Direct Inward System Access) Appli 0         
pbx_dundi.so                   Distributed Universal Number Discovery ( 0         
app_alarmreceiver.so           Alarm Receiver for Asterisk              0         
func_logic.so                  Logical dialplan functions               0         
format_gsm.so                  Raw GSM data                             0         
format_h264.so                 Raw H.264 data                           0         
app_db.so                      Database Access Functions                0         
res_convert.so                 File format conversion CLI command       0         
app_image.so                   Image Transmission Application           0         
codec_g726.so                  ITU G.726-32kbps G726 Transcoder         0         
codec_ulaw.so                  mu-Law Coder/Decoder                     0         
app_authenticate.so            Authentication Application               0         
chan_sip.so                    Session Initiation Protocol (SIP)        0         
app_voicemail_plain.so         Comedian Mail (Voicemail System)         0         
app_directory_plain.so         Extension Directory                      0         
format_g723.so                 G.723.1 Simple Timestamp File Format     0         
chan_iax2.so                   Inter Asterisk eXchange (Ver 2)          0         
format_pcm.so                  Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0         
func_callerid.so               Caller ID related dialplan functions     0         
func_timeout.so                Channel timeout dialplan functions       0         
app_privacy.so                 Require phone number to be entered, if n 0         
app_followme.so                Find-Me/Follow-Me Application            0         
app_getcpeid.so                Get ADSI CPE ID                          0         
app_morsecode.so               Morse code                               0         
app_waituntil.so               Wait until specified time                0         
codec_speex.so                 Speex Coder/Decoder                      0         
app_userevent.so               Custom User Event Application            0         
cdr_csv.so                     Comma Separated Values CDR Backend       0         
codec_adpcm.so                 Adaptive Differential PCM Coder/Decoder  0         
func_math.so                   Mathematical dialplan function           0         
func_sysinfo.so                System information related functions     0         
res_clioriginate.so            Call origination from the CLI            0         
app_record.so                  Trivial Record Application               0         
func_blacklist.so              Look up Caller*ID name/number from black 0         
chan_mgcp.so                   Media Gateway Control Protocol (MGCP)    0         
app_stack.so                   Dialplan subroutines (Gosub, Return, etc 0         
func_extstate.so               Gets an extension's state in the dialpla 0         
chan_oss.so                    OSS Console Channel Driver               0         
func_enum.so                   ENUM related dialplan functions          0         
func_devstate.so               Gets or sets a device state in the dialp 0         
app_forkcdr.so                 Fork The CDR into 2 separate entities    0         
format_sln16.so                Raw Signed Linear 16KHz Audio support (S 0         
func_shell.so                  Returns the output of a shell command    0         
chan_phone.so                  Linux Telephony API Support              0         
app_parkandannounce.so         Call Parking and Announce Application    0         
app_festival.so                Simple Festival Interface                0         
app_controlplayback.so         Control Playback Application             0         
func_md5.so                    MD5 digest dialplan functions            0         
app_cdr.so                     Tell Asterisk to not maintain a CDR for  0         
chan_agent.so                  Agent Proxy Channel                      0         
app_amd.so                     Answering Machine Detection Application  0         
codec_lpc10.so                 LPC10 2.4kbps Coder/Decoder              0         
app_talkdetect.so              Playback with Talk Detection             0         
func_version.so                Get Asterisk Version/Build Info          0         
148 modules loaded
Hate*CLI> 

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.

Thanx

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…

thanx

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!

===EDIT===
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