[Asterisk-Users] Forward incoming SIP calls to H323 ipphone?
Alex van Es
Alex at icepick.com
Mon Nov 8 11:23:03 MST 2004
Michael,
Attached some of the logging.
I noticed that when I call the sip number, it surely is talking to my
ipphone. When I look at the debug info coming out of my
phone it starts to spit out information (not readable) so for sure
asterisk and the phone are talking.
I tried setting a different codec in the oh323.conf, but that didn't
help..
Alex
Asterisk Ready.
*CLI> -- H.323 call to 192.168.1.20 with codec ALAW
Urgent handler
-- Called 192.168.1.20
Urgent handler
-- H.323 call 'ip$localhost/24187' cleared, reason 22 (Remote
endpoint is offline)
-- Hungup 'OH323/L24187'
== No one is available to answer at this time
Urgent handler
Asterisk Dynamic Loader Starting:
[chan_oh323.so] => (OpenH323 Channel Driver)
[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323
v1.13.5, PWlib v1.6.6
== OpenH323 Channel Ready (v0.6.3)
Nov 8 19:28:16 DEBUG[147465]: build_route: Record-Route hop:
<sip:495234 at 69.90.155.70;ftag=999500031330;lr=on>
Nov 8 19:28:16 DEBUG[147465]: build_route: Contact hop:
<sip:82.161.62.10:5060>
Nov 8 19:28:16 DEBUG[294930]: Launching 'System'
Nov 8 19:28:16 DEBUG[294930]: Launching 'System'
Nov 8 19:28:16 DEBUG[294930]: Launching 'Dial'
Nov 8 19:28:16 DEBUG[294930]: In oh323_request: type=OH323, format=8,
data=192.168.1.20.
Nov 8 19:28:16 DEBUG[294930]: Player fds 27,28 - Recorder fds 29,30 -
Event pipe 31,40.
Nov 8 19:28:16 DEBUG[294930]: Created new call structure 0 (5548
bytes).
Nov 8 19:28:16 DEBUG[294930]: OH323/L0: Raw format set to ALAW.
Nov 8 19:28:16 DEBUG[294930]: Context is 'voip-h323', extension is 's'.
Nov 8 19:28:16 DEBUG[294930]: CallerID/ANI is ''.
Nov 8 19:28:16 DEBUG[294930]: OH323/L0: Native format changed to ALAW.
Nov 8 19:28:16 DEBUG[294930]: In oh323_call (OH323/L0,
dest=192.168.1.20, timeout=0).
Nov 8 19:28:16 DEBUG[294930]: OH323/L0: Generating CallerID '"Alex"
<82.161.62.10>'
Nov 8 19:28:16 DEBUG[294930]: CID is '82.161.62.10'.
Nov 8 19:28:16 DEBUG[294930]: CIDname is 'Alex'.
Nov 8 19:28:16 DEBUG[294930]: OH323/192.168.1.20: No ${OH323_OUTCODEC}.
Nov 8 19:28:16 DEBUG[294930]: capability_set[0] - 2
Nov 8 19:28:16 DEBUG[294930]: capability_set[1] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[2] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[3] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[4] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[5] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[6] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[7] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[8] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[9] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[10] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[11] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[12] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[13] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[14] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[15] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[16] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[17] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[18] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[19] - 0
Nov 8 19:28:16 DEBUG[294930]: NEW STATE: NULL --> INIT
Nov 8 19:28:16 DEBUG[294930]: OH323/L24187: Call to 192.168.1.20
initiated successfully.
Nov 8 19:28:16 DEBUG[294930]: Set channel OH323/L24187 to read format
ULAW
Nov 8 19:28:16 DEBUG[294930]: Set channel SIP/55555-785e to write
format ULAW
Nov 8 19:28:16 DEBUG[294930]: Set channel OH323/L24187 to write format
ALAW
Nov 8 19:28:16 DEBUG[294930]: Set channel SIP/55555-785e to read
format ALAW
Nov 8 19:28:26 DEBUG[49155]: ENTER cleanup_h323_connection.
Nov 8 19:28:26 DEBUG[49155]: Call ip$localhost/24187 found in 0.
Nov 8 19:28:26 DEBUG[49155]: Call ip$localhost/24187 cleared in INIT
state.
Nov 8 19:28:26 DEBUG[49155]: NEW STATE: INIT --> CLEARED
Nov 8 19:28:26 DEBUG[49155]: Forcing H.323 channel to hangup.
Nov 8 19:28:26 DEBUG[294930]: OH323/L24187: Channel was shut down.
Nov 8 19:28:26 DEBUG[294930]: Hanging up channel 'OH323/L24187'
Nov 8 19:28:26 DEBUG[294930]: In oh323_hangup (OH323/L24187).
Nov 8 19:28:26 DEBUG[294930]: NEW STATE: CLEARED --> CLEARED
Nov 8 19:28:26 DEBUG[294930]: OH323/L24187: Call ip$localhost/24187
found in 0.
Nov 8 19:28:26 DEBUG[294930]: Releasing resources of call (0).
Nov 8 19:28:26 DEBUG[294930]: Releasing allocated resources (0).
Nov 8 19:28:26 DEBUG[294930]: Player fds 27,28 - Recorder fds 29,30 -
Event pipe 31,40.
Nov 8 19:28:26 DEBUG[294930]: Closing socket 28.
On 8-nov-04, at 18:53, Michael Manousos wrote:
>
> 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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>> --
>> Alex van Es - Alex at icepick.com
>> http://photography.icepick.com
>> ----------------------------------------------------------------------
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>
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--
Alex van Es - Alex at icepick.com
http://photography.icepick.com
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