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
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>
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…
Oh, ok, well i started the dahdi service and restarted asterisk, and now i get further along . 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).
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: