What to answer to this one? Module loaded and no 'OH323' channel type registered? How did you do that? As a last attempt, enable debugging on the console (logger.conf) and start Asterisk with -vvvcd, rerun and email the full output.

Also, send the portion of Asterisk boot messages (where it loads
the various modules) that belong to chan_oh323.so.

Michael.

Alex van Es wrote:
Michael,

When I do show modules it shows up in the list..
And if it wasn't loaded, how come asterisks can still receive h323 calls?

Alex


apeldoorn*CLI> show modules Module Description Use Count chan_modem.so Generic Voice Modem Driver 0 chan_modem_aopen.so A/Open (Rockwell Chipset) ITU-2 VoiceMod 0 res_musiconhold.so Music On Hold Resource 1 res_adsi.so ADSI Resource 1 res_features.so Call Parking Resource 1 res_crypto.so Cryptographic Digital Signatures 1 res_indications.so Indications Configuration 0 res_monitor.so Call Monitoring Resource 1 res_agi.so Asterisk Gateway Interface (AGI) 0 chan_sip.so Session Initiation Protocol (SIP) 0 chan_modem_bestdata.so BestData (Conexant V.90 Chipset) VoiceMo 0 chan_modem_i4l.so ISDN4Linux Emulated Modem Driver 0 chan_agent.so Agent Proxy Channel 0 chan_mgcp.so Media Gateway Control Protocol (MGCP) 0 chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 chan_local.so Local Proxy Channel 0 chan_skinny.so Skinny Client Control Protocol (Skinny) 0 chan_oss.so OSS Console Channel Driver 0 chan_phone.so Linux Telephony API Support 0 pbx_config.so Text Extension Configuration 0 pbx_wilcalu.so Wil Cal U (Auto Dialer) 0 pbx_spool.so Outgoing Spool Support 1 app_dial.so Dialing Application 0 app_playback.so Trivial Playback Application 0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_directory.so Extension Directory 0 app_mp3.so Silly MP3 Application 0 app_system.so Generic System() application 0 app_echo.so Simple Echo Application 0 app_record.so Trivial Record Application 0 app_image.so Image Transmission Application 0 app_url.so Send URL Applications 0 app_disa.so DISA (Direct Inward System Access) Appli 0 app_qcall.so Call from Queue 0 app_adsiprog.so Asterisk ADSI Programming Application 0 app_getcpeid.so Get ADSI CPE ID 0 app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 app_zapateller.so Block Telemarketers with Special Informa 0 app_setcallerid.so Set CallerID Application 0 app_festival.so Simple Festival Interface 0 app_queue.so True Call Queueing 0 app_senddtmf.so Send DTMF digits Application 0 app_parkandannounce.so Call Parking and Announce Application 0 app_striplsd.so Strip trailing digits 0 app_setcidname.so Set CallerID Name 0 app_lookupcidname.so Look up CallerID Name from local databas 0 app_substring.so (Deprecated) Save substring digits in a 0 app_macro.so Extension Macros 0 app_authenticate.so Authentication Application 0 app_softhangup.so Hangs up the requested channel 0 app_lookupblacklist.so Look up Caller*ID name/number from black 0 app_waitforring.so Waits until first ring after time 0 app_privacy.so Require phone number to be entered, if n 0 app_db.so Database access functions for Asterisk e 0 app_chanisavail.so Check if channel is available 0 app_enumlookup.so ENUM Lookup 0 app_transfer.so Transfer 0 app_setcidnum.so Set CallerID Number 0 app_cdr.so Make sure asterisk doesn't save CDR for 0 app_hasnewvoicemail.so Indicator for whether a voice mailbox ha 0 app_sayunixtime.so Say time 0 app_cut.so Cuts up variables 0 app_read.so Read Variable Application 0 app_setcdruserfield.so CDR user field apps 0 app_random.so Random goto 0 app_ices.so Encode and Stream via icecast and ices 0 app_eval.so Reevaluates strings 0 app_nbscat.so Silly NBS Stream Application 0 app_sendtext.so Send Text Applications 0 app_exec.so Executes applications 0 app_sms.so SMS/PSTN handler 0 app_groupcount.so Group Management Routines 0 app_txtcidname.so TXTCIDName 0 app_controlplayback.so Control Playback Application 0 app_talkdetect.so Playback with Talk Detection 0 app_alarmreceiver.so Alarm Receiver for Asterisk 0 app_userevent.so Custom User Event Application 0 app_verbose.so Send verbose output 0 app_test.so Interface Test Application 0 app_forkcdr.so Fork The CDR into 2 seperate entities. 0 codec_ilbc.so iLBC/PCM16 (signed linear) Codec Transla 0 codec_gsm.so GSM/PCM16 (signed linear) Codec Translat 0 codec_lpc10.so LPC10 2.4kbps (signed linear) Voice Code 0 codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 codec_ulaw.so Mu-law Coder/Decoder 0 codec_alaw.so A-law Coder/Decoder 0 codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_gsm.so Raw GSM data 0 format_wav.so Microsoft WAV format (8000hz Signed Line 0 format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 format_vox.so Dialogic VOX (ADPCM) File Format 0 format_pcm.so Raw uLaw 8khz Audio support (PCM) 0 format_g729.so Raw G729 data 0 format_pcm_alaw.so Raw aLaw 8khz PCM Audio support 0 format_h263.so Raw h263 data 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_ilbc.so Raw iLBC data 0 format_sln.so Raw Signed Linear Audio support (SLN) 0 format_jpeg.so JPEG (Joint Picture Experts Group) Image 0 cdr_csv.so Comma Separated Values CDR Backend 0 cdr_manager.so Asterisk Call Manager CDR Backend 0 chan_oh323.so OpenH323 Channel Driver 0 chan_zap.so Zapata Telephony 0 app_zapras.so Zap RAS Application 0 app_meetme.so MeetMe conference bridge 0 app_flash.so Flash zap trunk application 0 app_zapbarge.so Barge in on Zap channel application 0 app_zapscan.so Scan Zap channels application 0


On 8-nov-04, at 18:29, Michael Manousos wrote:

    Alex van Es wrote:

        Michael,
        Yeah.. for sure the channel is loaded.. calling to my asterisks
        works fine.
        I have included the oh323.conf and the original message.
        Thanks a lot for you help. I would would like to get this baby
        working.
        Alex
        The log;
        Nov 8 18:04:01 WARNING[294930]: channel.c:1901 ast_request: No
        channel type registered for 'OH323'


Hmm, according to this message, chan_oh323.so isn't loaded. Your config is fine.

    Michael.


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