[Asterisk-Users] Newbie needs help - Dev_Kit_Lite
installationproblem
David Luong
david.luong at embedia.com
Fri Aug 27 08:51:07 MST 2004
Thanks Don for the help but I found someone with a similar problem and
what they did was remove the audio module with a simple rmmod audio. Now
i get dial tone and everything from my both both my x100p and my s100u.
And can dial out from both.
Thanx again
Dave
>
> ----- Original Message -----
> From: "David Luong" <david.luong at embedia.com>
> To: <Asterisk-Users at lists.digium.com>
> Sent: Thursday, August 26, 2004 2:35 PM
> Subject: [Asterisk-Users] Newbie needs help - Dev_Kit_Lite
> installationproblem
>
>
>> Installing DevkitLite hardware (Very similar to John Lange's post on Tue
>> Oct 08 2002)
>>
>> I cannot get anything to work on the phone connected to the s100u. I
>> dont
>> know what to do.
>> Can someone please help me?
>
> I too had the DevKitLite hardware. Had nothing but problems with it. Is
> difficult to get running and when you do get it going, I think you will
> find
> that you may have trouble getting it to dial correctly. After getting it
> running, I would pick up the phone to make a call and I would get a dial
> tone but the S100U would not accept the DTMF tones for dialing. The only
> thing I could do to correct this would be to down * and reboot the
> computer.
> Finally I stopped using it and bought a one port TDM400U. It works with
> no
> problems.
>>
>> I used the sample configuration files from digium documentaion that was
>> supposed to be "sane" defaults for the kit.
>>
>> Very similar to John Lange's post on Tue Oct 08 2002
>> Here is my probelm:
>> This is what i did.
>> unplugged s100u
>> rmmod wcfxo
>> rmmod wcusb
>> rmmod zaptel
>> replugged s100u
>> modprobe wcfxo
>> modprobe wcusb
>> ztcfg -vv
>> asterisk -cv
>> This is what I got:
>>
>> [root at localhost root]# modprobe wcfxo
>> ZT_CHANCONFIG failed on channel 2: No such device or address (6)
>> /lib/modules/2.4.20-8/misc/wcfxo.o: post-install wcfxo failed
>> /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed
>> [root at localhost root]# modprobe wcusb
>> [root at localhost root]# ztcfg -vv
>>
>> Zaptel Configuration
>> ======================
>>
>>
>> Channel map:
>>
>> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
>> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
>>
>> 2 channels configured.
>>
>> ZT_CHANCONFIG failed on channel 2: No such device or address (6)
>>
>> [root at localhost root]# asterisk -cv
>> Asterisk CVS-HEAD-08/24/04-09:05:32, Copyright (C) 1999-2004 Digium.
>> Written by Mark Spencer <markster at digium.com>
>> =========================================================================
>> Asterisk Event Logger Started /var/log/asterisk/event_log
>> Asterisk PBX Core Initializing
>> Registering builtin applications:
>> [AbsoluteTimeout]
>> [Answer]
>> [BackGround]
>> [Busy]
>> [Congestion]
>> [DigitTimeout]
>> [Goto]
>> [GotoIf]
>> [GotoIfTime]
>> [Hangup]
>> [NoOp]
>> [Prefix]
>> [Progress]
>> [ResetCDR]
>> [ResponseTimeout]
>> [Ringing]
>> [SayNumber]
>> [SayDigits]
>> [SayAlpha]
>> [SayPhonetic]
>> [SetAccount]
>> [SetAMAFlags]
>> [SetGlobalVar]
>> [SetLanguage]
>> [SetVar]
>> [StripMSD]
>> [Suffix]
>> [Wait]
>> [WaitExten]
>> Asterisk Dynamic Loader Starting:
>> [chan_modem.so] => (Generic Voice Modem Driver)
>> => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver)
>> [res_musiconhold.so] => (Music On Hold Resource)
>> [res_adsi.so] => (ADSI Resource)
>> [res_features.so] => (Call Parking Resource)
>> [res_crypto.so] => (Cryptographic Digital Signatures)
>> [res_indications.so] => (Indications Configuration)
>> [res_monitor.so] => (Call Monitoring Resource)
>> [res_agi.so] => (Asterisk Gateway Interface (AGI))
>> [chan_sip.so] -z: No such file or directory
>> => (Session Initiation Protocol (SIP))
>> [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset)
>> VoiceModem
>> Driver)
>> [chan_modem_i4l.so] => (ISDN4Linux Emulated Modem Driver)
>> [chan_agent.so] => (Agent Proxy Channel)
>> [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP))
>> [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
>> [chan_local.so] => (Local Proxy Channel)
>> [chan_skinny.so] => (Skinny Client Control Protocol (Skinny))
>> Aug 26 15:12:15 WARNING[1076220544]: chan_skinny.c:2584 reload_config:
>> Unable to get our IP address,
>> Skinny disabled
>> [chan_oss.so] => (OSS Console Channel Driver)
>> Aug 26 15:12:15 WARNING[1076220544]: chan_oss.c:992 load_module: XXX I
>> don't work right with non-full duplex sound cards XXX
>> Aug 26 15:12:15 WARNING[1097410752]: chan_oss.c:239 sound_thread: Read
>> error on sound device: Resource temporarily unavailable
>> [chan_phone.so] => (Linux Telephony API Support)
>> [chan_zap.so] => (Zapata Telephony w/PRI)
>> Aug 26 15:12:16 WARNING[1076220544]: chan_zap.c:721 zt_open: Unable to
>> specify channel 2: Device or resource busy
>> Aug 26 15:12:16 ERROR[1076220544]: chan_zap.c:5869 mkintf: Unable to
>> open
>> channel 2: Device or resource busy
>> here = 0, tmp->channel = 2, channel = 2
>> Aug 26 15:12:16 ERROR[1076220544]: chan_zap.c:8776 setup_zap: Unable to
>> register channel '2'
>> Aug 26 15:12:16 WARNING[1076220544]: loader.c:328 ast_load_resource:
>> chan_zap.so: load_module failed, returning -1
>> -- Unregistered channel 1
>> Aug 26 15:12:16 WARNING[1076220544]: loader.c:423 load_modules: Loading
>> module chan_zap.so failed!
>>
>> A "ps" shows that asterisk didn't start. And there is still no dialtone
>> from the phone on the s100u. I dont' know what to do. My x100p is
>> plugged into a open pci slot with the phone line from the wall into the
>> "wall" jack, and a normal cheapo storebought phone connected to the
>> phone
>> slot. I have my s100u plugged in and nother cheapo phone connected to
>> that as well.
>> I get a dialtone from my phone connected to my x100p with or without
>> asterisk(I can call out and received calls like as if it was plugged
>> straight into the wall and not through the x100p)
>> I'm running Redhat9.0 on a P2 300Mhz emachine
>> some more details:
>>
>> lsmod gives me:
>> [root at localhost root]# lsmod
>> Module Size Used by Not tainted
>> wcusb 20064 0 (unused)
>> wcfxo 9376 0 (unused)
>> zaptel 178080 0 [wcusb wcfxo]
>> cs46xx 62832 0 (autoclean)
>> ac97_codec 13640 0 (autoclean) [cs46xx]
>> parport_pc 19076 1 (autoclean)
>> lp 8996 0 (autoclean)
>> parport 37056 1 (autoclean) [parport_pc lp]
>> autofs 13268 0 (autoclean) (unused)
>> tulip 43840 1
>> sg 36524 0 (autoclean)
>> sr_mod 18136 0 (autoclean)
>> ide-scsi 12208 0
>> scsi_mod 107160 3 [sg sr_mod ide-scsi]
>> ide-cd 35708 0
>> cdrom 33728 0 [sr_mod ide-cd]
>> audio 46648 0 (unused)
>> soundcore 6404 4 [cs46xx audio]
>> keybdev 2944 0 (unused)
>> mousedev 5492 1
>> hid 22148 0 (unused)
>> input 5856 0 [keybdev mousedev hid]
>> usb-uhci 26348 0 (unused)
>> usbcore 78784 1 [wcusb audio hid usb-uhci]
>> ext3 70784 2
>> jbd 51892 2 [ext3]
>>
>> my /etc/zaptel.conf is as follows:
>>
>> fxsks=1
>> fxoks=2
>> loadzone = us
>> defaultzone=us
>>
>> my /etc/asterisk/zapata.conf is(not including edited out lines):
>> [channels]
>> callwaiting=yes
>> callwaitingcallerid=yes
>> threewaycalling=yes
>> transfer=yes
>> cancallforward=yes
>> rxgain=0.0
>> txgain=0.0
>> group=1
>> immediate=no
>> context=bell
>> signalling=fxs_ks
>> channel=1
>> context=home
>> signalling=fxo_ks
>> channel=2
>>
>> I emailed support at digium and they told me to switch chanel 1 and channel
>> 2
>> in zaptel.conf. and to change the fxo_ks and fxs_ks in zapata.conf.
>> and
>> then doing the modprobes with wcusb first. also didn't work. And
>> besides
>> I read that it's better to have the x100p as channel 1 and the usb for
>> channel 2.
>
> I agree that swapping the channels and the signalling should make it work.
> It did for me. Remember to get your signalling for each channel the same
> in
> both zaptel.conf and zapata.conf the same. It will not work if they are
> different. Also make sure the extensions.conf has the correct channels in
> the extensions. Try rebooting your computer when you change this
> information instead of rmmoding the modules and reloading. The rmmoding
> did
> not work for me, but rebooting did.
>
>>
>> Anyways I thank anybody and everybody that took the time and even
>> considered helping a Asterisk Newbie
>>
>> Dave
>>
>>
>>
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>
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