[Asterisk-Dev] X100P doesn`t call
Sergio Serrano Revuelto
sergio.serrano at avanzada7.com
Thu Apr 10 13:45:38 MST 2003
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?
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
>
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