[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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