[Asterisk-Users] Forward incoming SIP calls to H323 ipphone?

Alex van Es Alex at icepick.com
Mon Nov 8 10:42:47 MST 2004


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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--
Alex van Es - Alex at icepick.com
http://photography.icepick.com
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