[Asterisk-Dev] X100P doesn`t call
Michael Manousos
manousos at inaccessnetworks.com
Mon Apr 14 05:12:59 MST 2003
Sergio Serrano Revuelto wrote:
> I give you more information.
>
> Phone IP <--> CHAN_SIP <--> ASTERISK <--> X100P
>
> All work fine
>
> Phone IP <--> CHAN_OH323 <--> ASTERISK <--> X100P
> I have the problem described below.
> What's wrong?
Try the previous version of asterisk-oh323 (0.5.0).
There are some problems with the handling of
multiple frames/RTP packet in 0.5.1.
Michael Manousos.
> Could be alaw to slinear translator?
>
>
> Thanks in advance
> srsergio
>
> -----Mensaje original-----
> De: asterisk-dev-admin at lists.digium.com
> [mailto:asterisk-dev-admin at lists.digium.com] En nombre de Sergio Serrano
> Revuelto
> Enviado el: jueves, 10 de abril de 2003 19:53
> Para: asterisk-dev at lists.digium.com
> Asunto: RE: [Asterisk-Dev] X100P doesn`t call
>
>
> Thanks, you have reason. It was a problem with via C3 bios. Once I have
> reolved this problem, I have another problem. When I call in the next
> scenario:
>
> Phone IP <--> CHAN_OH323 <--> ASTERISK <--> X100P
>
> In The direction
> Phone IP <-- CHAN_OH323 <-- ASTERISK <-- X100P
> All work well, but in the other direction I hear only noise.
>
> What's wrong?
>
> My oh323.conf:
>
> [general]
> listenAddress=0.0.0.0
> listenPort=1720
> connectPort=1720
> fastStart=no
> h245Tunnelling=yes
> h245inSetup=no
> inBandDTMF=no
> silenceSuppression=no
> jitterMin=20
> jitterMax=100
> ipTos=none
> outboundMax=10
> inboundMax=10
> gatekeeper=192.168.0.204
> userInputMode=STRING
> context=voip-h323
> [register]
> alias=nbx1
> gwprefix=7
> gwprefix=9
> [codecs]
> codec=G711A
> frames=10
>
>
> My zapata.conf:
>
> [channels]
> language=en
> context=outgoing
> signalling=fxs_ks
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> threewaycalling=yes
> transfer=yes
> echocancel=16
> channel => 1
>
> Thanks in advance
> srsergio
>
>
>
>
> -----Mensaje original-----
> De: asterisk-dev-admin at lists.digium.com
> [mailto:asterisk-dev-admin at lists.digium.com] En nombre de Mark Spencer
> Enviado el: jueves, 10 de abril de 2003 17:33
> Para: asterisk-dev at lists.digium.com
> Asunto: RE: [Asterisk-Dev] X100P doesn`t call
>
>
> That usually means your card isn't taking interrupts for some reason.
>
> Mark
>
> On Thu, 10 Apr 2003, Sergio Serrano Revuelto wrote:
>
>
>>I'll found that proble isn't there. I'm sorry. Proble is motherboard
>>that I'm using. I have a mini-ITX with only a PCI slot. When I exec
>>modprobe wcfxo system says the next:
>>
>>Wcfxo: out of space to write register 06 with e0
>>Failed to initailize DAA, giving up...
>>
>>Someone knows something about this?
>>
>>Thanks in advance
>>srsergio
>>
>>
>>
>>-----Mensaje original-----
>>De: asterisk-dev-admin at lists.digium.com
>>[mailto:asterisk-dev-admin at lists.digium.com] En nombre de Sergio
>>Serrano Revuelto Enviado el: jueves, 10 de abril de 2003 13:45
>>Para: asterisk-dev at lists.digium.com
>>Asunto: [Asterisk-Dev] X100P doesn`t call
>>
>>
>>I have to upgrading all source code:
>> PWLIB, OPENH323, OPENH323GK, ZAPTEL, ZAPATA, ASTERISK;
>
> ASTERISK-OH323
>
>>and I begin from zero.
>>
>>When I call intrenally all go well, but when I call out, call doesn't
>>progress.
>>
>>Call start here
>> == Accepting call on 'H323:24943' (12)
>> -- Executing StripMSD("H323:24943", "1") in new stack
>> -- Executing Dial("H323:24943", "Zap/1/BYEXTENSION") in new stack
>> -- Called 1/637591599
>>
>>Call stop here and it doesn't progress
>>
>>When I hungup trace continue
>>
>> -- Hungup 'Zap/1-1'
>> == Spawn extension (outgoing, 637591599, 2) exited non-zero on
>>'H323:24943'
>> -- Hungup 'H323:24943'
>>
>>
>>
>>My extensions.conf is the next:
>>
>>[default]
>>exten => i,1,Playback,invalid
>>exten => s,1,Wait,1
>>exten => s,2,Answer
>>exten => s,3, SetMusicOnHold,default
>>exten => s,4,Goto,voip-h323|712|1
>>
>>[outgoing]
>>exten => _9XXXXXXXXX,1,Stripmsd,1
>>exten => _XXXXXXXXX,2,Dial,Zap/1/BYEXTENSION
>>
>>[voip-h323]
>>include => outgoing
>>exten => 712,1,Wait,1
>>exten => 712,2,Dial,OH323/12|20|mi|t|T
>>exten => 723,1,Wait,1
>>exten => 723,2,Dial,OH323/23|20|m|t|T
>>exten => 731,1,Dial,IAX/nbx1/712 at voip-h323
>>
>>
>>
>>
>>The booting of asterisk is the next.
>>
>> == Parsing '/etc/asterisk/asterisk.conf': Found
>>Asterisk 0.3.0, Copyright (C) 1999-2001 Linux Support Services, Inc.
>>Written by Mark Spencer <markster at linux-support.net>
>>======================================================================
>>==
>>=
>> == Parsing '/etc/asterisk/logger.conf': Found
>>Asterisk Event Logger Started /var/log/asterisk/event_log
>> == Manager registered action Ping
>> == Manager registered action Logoff
>> == Manager registered action Hangup
>> == Manager registered action Status
>> == Manager registered action Redirect
>> == Manager registered action Originate
>> == Manager registered action Command
>> == Parsing '/etc/asterisk/manager.conf': Found
>>Asterisk Management interface listening on port 5038
>>Asterisk PBX Core Initializing
>>Registering builtin applications:
>> [Answer]
>> == Registered application 'Answer'
>> [Goto]
>> == Registered application 'Goto'
>> [Hangup]
>> == Registered application 'Hangup'
>> [DigitTimeout]
>> == Registered application 'DigitTimeout'
>> [ResponseTimeout]
>> == Registered application 'ResponseTimeout'
>> [AbsoluteTimeout]
>> == Registered application 'AbsoluteTimeout'
>> [BackGround]
>> == Registered application 'BackGround'
>> [Wait]
>> == Registered application 'Wait'
>> [StripMSD]
>> == Registered application 'StripMSD'
>> [Prefix]
>> == Registered application 'Prefix'
>> [SetLanguage]
>> == Registered application 'SetLanguage'
>> [Ringing]
>> == Registered application 'Ringing'
>> [Congestion]
>> == Registered application 'Congestion'
>> [Busy]
>> == Registered application 'Busy'
>> [Setvar]
>> == Registered application 'Setvar'
>> [GotoIf]
>> == Registered application 'GotoIf'
>>Asterisk Dynamic Loader Starting:
>> == Parsing '/etc/asterisk/modules.conf': Found [res_musiconhold.so]
>>=> (Music On Hold Resource)
>> == Parsing '/etc/asterisk/musiconhold.conf': Found
>> == Registered application 'MusicOnHold'
>> == Registered application 'WaitMusicOnHold'
>> == Registered application 'SetMusicOnHold'
>> [res_adsi.so] => (Call Parking Resource)
>> == Parsing '/etc/asterisk/adsi.conf': Found
>> [res_parking.so] => (Call Parking Resource)
>> == Parsing '/etc/asterisk/parking.conf': Found
>> -- Registered extension context 'parkedcalls'
>> -- Added extension '701' priority 1 to parkedcalls
>> -- Added extension '702' priority 1 to parkedcalls
>> -- Added extension '703' priority 1 to parkedcalls
>> -- Added extension '704' priority 1 to parkedcalls
>> -- Added extension '705' priority 1 to parkedcalls
>> -- Added extension '706' priority 1 to parkedcalls
>> -- Added extension '707' priority 1 to parkedcalls
>> -- Added extension '708' priority 1 to parkedcalls
>> -- Added extension '709' priority 1 to parkedcalls
>> -- Added extension '710' priority 1 to parkedcalls
>> -- Added extension '711' priority 1 to parkedcalls
>> -- Added extension '712' priority 1 to parkedcalls
>> -- Added extension '713' priority 1 to parkedcalls
>> -- Added extension '714' priority 1 to parkedcalls
>> -- Added extension '715' priority 1 to parkedcalls
>> -- Added extension '716' priority 1 to parkedcalls
>> -- Added extension '717' priority 1 to parkedcalls
>> -- Added extension '718' priority 1 to parkedcalls
>> -- Added extension '719' priority 1 to parkedcalls
>> -- Added extension '720' priority 1 to parkedcalls
>>Junk at the beginning 49443303
>>Junk at the beginning 49443303
>>Warning, flexibel rate not heavily tested!
>>Warning, flexibel rate not heavily tested!
>> == Registered application 'ParkedCall'
>> [res_crypto.so] => (Cryptographic Digital Signatures)
>> -- Loaded PUBLIC key 'iaxtel'
>> [res_indications.so] => (Indications Configuration)
>> == Parsing '/etc/asterisk/indications.conf': Found
>> -- Registered indication country 'uk'
>> -- Registered indication country 'de'
>> -- Registered indication country 'nl'
>> -- Registered indication country 'fr'
>> -- Registered indication country 'au'
>> -- Registered indication country 'us'
>> -- Setting default indication country to 'us'
>> == Registered application 'Playtones'
>> == Registered application 'StopPlaytones'
>> [skipping chan_modem.so]
>> [chan_iax.so] => (Inter Asterisk eXchange)
>> == Manager registered action IAXpeers
>> == Parsing '/etc/asterisk/iax.conf': Found
>> == Registered channel type 'IAX' (Inter Asterisk eXchange Drver)
>> == Using TOS bits 16
>>Junk at the beginning 49443303
>> == IAX Ready and Listening on 192.168.0.204 port 5036 [skipping
>>chan_sip.so] [skipping chan_modem_aopen.so] [skipping chan_oss.so]
>>[skipping chan_modem_bestdata.so] [skipping chan_modem_i4l.so]
>>[chan_agent.so] => (Agent Proxy Channel)
>> == Registered channel type 'Agent' (Call Agent Proxy Channel)
>> == Registered application 'AgentLogin'
>> == Parsing '/etc/asterisk/agents.conf': Found
>> [skipping chan_mgcp.so]
>> [chan_phone.so] => (Linux Telephony API Support)
>> == Parsing '/etc/asterisk/phone.conf': Found
>> == Registered channel type 'Phone' (Standard Linux Telephony API
>>Driver)
>> [chan_zap.so] => (Zapata Telephony)
>> == Parsing '/etc/asterisk/zapata.conf': Found
>> -- Registered channel 1, FXS Kewlstart signalling
>> == Registered channel type 'Zap' (Zapata Telephony Driver)
>> == Registered channel type 'Tor' (Zapata Telephony Driver) Warning,
>>flexibel rate not heavily tested! [pbx_config.so] => (Text Extension
>>Configuration)
>> == Parsing '/etc/asterisk/extensions.conf': Found
>> -- Registered extension context 'voip-h323'
>> -- Including context 'outgoing' in context 'voip-h323'
>> -- Added extension '712' priority 1 to voip-h323
>> -- Added extension '712' priority 2 to voip-h323
>> -- Added extension '723' priority 1 to voip-h323
>> -- Added extension '723' priority 2 to voip-h323
>> -- Added extension '731' priority 1 to voip-h323
>> -- Registered extension context 'outgoing'
>> -- Added extension '_9XXXXXXXXX' priority 1 to outgoing
>> -- Added extension '_XXXXXXXXX' priority 2 to outgoing
>> -- Registered extension context 'default'
>> -- Added extension 'i' priority 1 to default
>> -- Added extension 's' priority 1 to default
>> -- Added extension 's' priority 2 to default
>> -- Added extension 's' priority 3 to default
>> -- Added extension 's' priority 4 to default [pbx_wilcalu.so] =>
>>(Wil Cal U (Auto Dialer)) [pbx_spool.so] => (Outgoing Spool Support)
>>/var/spool/asterisk/outgoing [app_dial.so] => (Dialing Application)
>> == Registered application 'Dial'
>> [app_playback.so] => (Trivial Playback Application)
>> == Registered application 'Playback'
>> [app_voicemail.so] => (Comedian Mail (Voicemail System))
>> == Registered application 'VoiceMail'
>> == Registered application 'VoiceMailMain'
>> [app_directory.so] => (Extension Directory)
>> == Registered application 'Directory'
>> [skipping app_intercom.so]
>> [app_mp3.so] => (Silly MP3 Application)
>> == Registered application 'MP3Player'
>> [app_system.so] => (Generic System() application)
>> == Registered application 'System'
>> [app_echo.so] => (Simple Echo Application)
>> == Registered application 'Echo'
>> [app_record.so] => (Trivial Record Application)
>> == Registered application 'Record'
>> [app_image.so] => (Image Transmission Application)
>> == Registered application 'SendImage'
>> [app_url.so] => (Send URL Applications)
>> == Registered application 'SendURL'
>> [app_disa.so] => (DISA (Direct Inward System Access) Application)
>> == Registered application 'DISA'
>> [app_agi.so] => (Asterisk Gateway Interface (AGI))
>> == Registered application 'AGI'
>> [app_qcall.so] => (Call from Queue)
>> [app_adsiprog.so] => (Asterisk ADSI Programming Application)
>> == Registered application 'ADSIProg'
>> [app_getcpeid.so] => (Get ADSI CPE ID)
>> == Registered application 'GetCPEID'
>> [app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application)
>> == Registered application 'Milliwatt'
>> [app_zapateller.so] => (Block Telemarketers with Special Information
>>Tone)
>> == Registered application 'Zapateller'
>> [app_datetime.so] => (Date and Time)
>> == Registered application 'DateTime'
>> [app_setcallerid.so] => (Set CallerID Application)
>> == Registered application 'SetCallerID'
>> [app_festival.so] => (Simple Festival Interface)
>> == Registered application 'Festival'
>> [app_queue.so] => (True Call Queueing)
>> == Registered application 'Queue'
>> == Manager registered action Queues
>> == Parsing '/etc/asterisk/queues.conf': Found [app_senddtmf.so] =>
>>(Send DTMF digits Application)
>> == Registered application 'SendDTMF'
>> [app_striplsd.so] => (Strip trailing digits)
>> == Registered application 'StripLSD'
>> [app_parkandannounce.so] => (Call Parking and Announce Application)
>> == Registered application 'ParkAndAnnounce' [app_setcidname.so] =>
>>(Set CallerID Name)
>> == Registered application 'SetCIDName'
>> [app_lookupcidname.so] => (Look up CallerID Name from local database)
>> == Registered application 'LookupCIDName'
>> [app_substring.so] => (Save substring digits in a given variable)
>> == Registered application 'SubString'
>> [app_zapras.so] => (Zap RAS Application)
>> == Registered application 'ZapRAS'
>> [app_meetme.so] => (Simple MeetMe conference bridge)
>> == Registered application 'MeetMeCount'
>> == Registered application 'MeetMe'
>> [app_flash.so] => (Flash zap trunk application)
>> == Registered application 'Flash'
>> [app_zapbarge.so] => (Barge in on Zap channel application)
>> == Registered application 'ZapBarge'
>> [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator)
>> == Registered translator 'gsmtolin' from format 1 to 6, cost 6
>> == Registered translator 'lintogsm' from format 6 to 1, cost 26
>>[codec_mp3_d.so] => (MP3/PCM16 (signed linear) Translator (Decoder
>>only))
>> == Registered translator 'mp3tolin' from format 4 to 6, cost 52
>>[codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder)
>> == Registered translator 'lpc10tolin' from format 7 to 6, cost 48
>> == Registered translator 'lintolpc10' from format 6 to 7, cost 85
>>[codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder)
>> == Registered translator 'adpcmtolin' from format 5 to 6, cost 2
>> == Registered translator 'lintoadpcm' from format 6 to 5, cost 3
>>[codec_ulaw.so] => (Mu-law Coder/Decoder)
>> == Registered translator 'ulawtolin' from format 2 to 6, cost 1
>> == Registered translator 'lintoulaw' from format 6 to 2, cost 1
>>[codec_alaw.so] => (A-law Coder/Decoder)
>> == Registered translator 'alawtolin' from format 3 to 6, cost 1
>> == Registered translator 'lintoalaw' from format 6 to 3, cost 1
>>[codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder)
>> == Registered translator 'alawtoulaw' from format 3 to 2, cost 1
>> == Registered translator 'ulawtoalaw' from format 2 to 3, cost 1
>>[format_g723.so] => (G.723.1 Simple Timestamp File Format)
>> == Registered file format g723sf, extension(s) g723 [format_wav.so]
>>=> (Microsoft WAV format (8000hz Signed Linear))
>> == Registered file format wav, extension(s) wav [format_mp3.so] =>
>>(MPEG-1,2 Layer 3 File Format Support)
>> == Registered file format mp3, extension(s) mp3|mpeg3
>>[format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM))
>> == Registered file format wav49, extension(s) WAV [format_gsm.so]
>
> =>
>
>>(Raw GSM data)
>> == Registered file format gsm, extension(s) gsm [format_vox.so] =>
>>(Dialogic VOX (ADPCM) File Format)
>> == Registered file format vox, extension(s) vox [format_pcm.so] =>
>>(Raw uLaw 8khz Audio support (PCM))
>> == Registered file format pcm, extension(s) pcm|ulaw|ul|mu
>>[format_g729.so] => (Raw G729 data)
>> == Registered file format g729, extension(s) g729 [format_jpeg.so]
>
> =>
>
>>(JPEG (Joint Picture Experts Group) Image Format)
>> == Registered format 'jpg' (JPEG (Joint Picture Experts Group))
>>[cdr_csv.so] => (Comma Separated Values CDR Backend) [cdr_mysql.so]
>
> =>
>
>>(MySQL CDR Backend)
>> == Parsing '/etc/asterisk/cdr_mysql.conf': Found [chan_oh323.so] =>
>
>
>>(OpenH323 Channel Driver)
>> == Parsing '/etc/asterisk/oh323.conf': Found
>> == Registered channel type 'OH323' (OpenH323 Channel Driver)
>> == OpenH323 Channel Ready (v0.5.1)
>>Asterisk Ready.
>>
>>
>>
>>
>>Could someone help me? I'm newbie
>>
>>Thanks in Advance
>>srsergio
>>
>>_______________________________________________
>>Asterisk-Dev mailing list
>>Asterisk-Dev at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>>_______________________________________________
>>Asterisk-Dev mailing list
>>Asterisk-Dev at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>
>
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