Hello,
I have purchased a Mobigater Mini, and have spent 50+hours trying to get this installed and working with the celliax module in asterisk 1.4. There is NO documentation anywhere, except how to build the module, which was poor at best.
I am really dissappointed with the lack of forum activity, and on top of that the old forum which has EXACTLY what I need is gone!
Don't get me wrong, I appreciate (greatly) all of the work that you folks are doing. But what good is it if there is no documentation.
The said thing is that I am a newb when it comes to linux and channel drivers, and I know that with just a little guidance I would be up and running in just a couple of minutes.
Is there someone that can and would be willing to help me? I am willing to offer a $50 bounty for someone to assist me with this.
what is your problem?
what is your problem chuck?
chan_celliax is used like any other channel in asterisk
try to be more specific, thanks
also, you wrote on saturday and sunday, and you got answer.
If your questions are about how to use trixbox or asterisk, or how to add a channel to trixbox, those questions are better suited for asterisk or trixbox forums.
That said, please feel free to ask questions that are specific to chan_celliax.
Thanks for your appreciation of the effort we put in free software.
starting on asterisk
I don't use trixbox, so I can give you hints on how to use chan_celliax on asterisk:
copy the celliax.conf file where the other configuration files of asterisk are
copy chan_celliax.so where the other asterisk modules are
start asterisk with -c option (or connect to the running asterisk with "asterisk -r")
type from the asterisk command line:
set debug 100
load chan_celliax
then if you have problems, please copy and paste the results here
Thanks for your responses!!
Wonderful!
I would LOVE some help.
At this point I have configured asterisk to load chan_celliax.so on startup. This is the first of the errors that I am seeing:
Jan 17 16:33:42] DEBUG[12070] chan_celliax.c: rev exported[(nil)|b7fd58d0][DEBUG_SERIAL 2624 ][line0 ][-1, 0, 0] serial error: Permission denied
[Jan 17 16:33:42] ERROR[12070] chan_celliax.c: rev exported[(nil)|b7fd58d0][ERROR 1989 ][line0 ][-1, 0, 0] celliax_serial_init failed
[Jan 17 16:33:42] ERROR[12070] chan_celliax.c: rev exported[(nil)|b7fd58d0][ERROR 1350 ][none ][-1,-1,-1] Unable to create channel Celliax from celliax.conf category '[line0]'
[Jan 17 16:33:42] VERBOSE[12070] logger.c: == Unregistered channel type 'Celliax'
Any thoughts? (PS. I will be happy to provide the $50, it wasnt just an attention getter, I really do appreciate your help!)
no need for bounties, thx ;)
:)
no need for bounties, just ask and be patient :)
as the error says, the user asterisk is running as is not getting permissions to use the serial devices
try to execute this commands from the linux (not asterisk) command line, after logging in as root:
chmod -R a+rw /dev/snd
chmod -R a+rw /dev/ttyA*
then from the asterisk command line:
set debug 100
load chan_celliax.so
No Channels available..
Ok,
threw this command at it:
chown -R asterisk:asterisk /dev/ttyACM0
Took care of that error. There does not seem to be an error now.
However, when I use this Dial string:
Celliax/line0/$OUTNUM$
It always says no channels available.
good!
good!
can you copy here your celliax.conf file?
please copy here the debug
after having typed at the asterisk command line:
set debug 100
try to make the call and copy here all the output
celliax.conf
Thanks again for your help!
;;
;;
; Celliax Asterisk Driver
;
; Configuration file
; lines beginning with semicolon (" are ignored (commented out)
;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;
; The first interface (named line0)
[line0]
;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; general settings, valid on all platforms
;
;
; Default language
;
language=en
;
; Default context (is overridden with @context syntax)
;
context=default
;
; Default extension (in extensions.conf) where incoming calls land
;
extension=s
;
;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; Debugging settings, valid globally for all interfaces on all platforms
;
; the debug values are global for all the interfaces.
;
; default is no celliax debugging output, you **have** to activate debugging here to obtain debugging from celliax
;
; To see the debugging output you have to "set debug 100" from the Asterisk CLI or launch
; Asterisk with -ddddddddddd option, and have the logger.conf file activating debug info for console and messages
;
; You can activate each of the following separately, but you can't disactivate. Eg: debug_at=no does not subtract debug_at from debug_all
; debug_all activate all possible debugging info
;
;debug_all=yes
debug_at=yes
;debug_fbus2=yes
debug_serial=yes
debug_pbx=yes
debug_sound=yes
;debug_locks=yes
debug_call=yes
;debug_monitorlocks=yes
;debug_cvm=yes
;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; This is the program that will receive in stdin the incoming SMSs
sms_receiving_program=/usr/local/asterisk/usr/sbin/ciapalo
;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; serial settings, valid for all platforms
;
;control_device_protocol can be AT or FBUS2 or NO_SERIAL (with NO_SERIAL the speed and name of the port are ignored)
control_device_protocol=at
;speed of the serial port
control_device_speed=115200
;name of the serial port device
control_device_name=/dev/ttyACM0 ; this is a Celliax Official Device, recognized as a modem by Linux
;control_device_name=/dev/ttyUSB0 ; this is an alternative form of a Celliax Official Device, recognized as a modem by Linux
;watch the soundcard for noise (ring), because the serial port do not tell us about incoming calls (eg 3310nokia), NO_SERIAL protocol watch for acoustic ring in any case
need_acoustic_ring=0
;audio noise threshold beyond which we declare there is a ring (512 is default, put it to 1024 or 2048 if you have false positive), ignored if not watching for ring
dsp_silence_threshold=1024
;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; audio boost settings, valid for all platforms, to compensate for different soundcard/phone input/output signal levels
; tweak it if you get horrible (or not hearable) sound
;
;boost can be positive or negative (-40 to +40) in db
;experiment to find which values are best for your soundcard
playback_boost=0 ;
capture_boost=0 ;
;;;;;;;;;;;;;;;;;;;;;;;;;;;
; which audio device to use
;;;;;;;;;;;;;;;;;;;;;;;;;;;
;names of the sound devices in linux
;if you don't use skype on this interface (eg don't need to share the audio device with other applications while celliax is running), use the plughw:n devices (plughw:0 is the first, plughw:1 is the second soundcard, etc). They have the best latency
;if you use skype on this interface use the default:n devices (default:0 is the first, default:1 is the second soundcard, etc). They have worst latency, but you can share them
alsa_capture_device_name=plughw:1
alsa_playback_device_name=plughw:1
alsa_period_size=160
alsa_periods_in_buffer=4
;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; at "modem" commands settings for this interface (if controldevice_protocol is not AT they are ignored)
;
;what the modem is expecting in the part of the dial command before the number to be dialed (eg: ATD)
;at_dial_pre_number=AT+CKPD="EEE
at_dial_pre_number=ATD
;what the modem is expecting in the part of the dial command after the number to be dialed. If you want it to wait for a semicolon (;), just comment out the followin line. Wait for semicolon is the default
;at_dial_post_number=S"
;what the modem will answer after succesful execution of the dial command
at_dial_expect=OK
;command to hangup the current call
;at_hangup=AT+CKPD="EEE"
at_hangup=ATH
;what the modem will answer after succesful execution of the hangup command
at_hangup_expect=OK
;command to answer an incoming call
at_answer=ATA
;what the modem will answer after succesful execution of the answer command
at_answer_expect=OK
;pause right after serial port opening, before any command is sent, in usecs (1million usec= 1sec)
at_initial_pause=500000
;custom commands to be sent after the initial pause and before the "built in" initialization commands, and what the modem is expected to send as reply
;the first empty string stop the preinit sending
at_preinit_1=atciapa ; nonsense entry, just to show the preinit
at_preinit_1_expect=OK
at_preinit_2=
at_preinit_2_expect=
at_preinit_3=
at_preinit_3_expect=
at_preinit_4=
at_preinit_4_expect=
at_preinit_5=
at_preinit_5_expect=
;pause right after the custom preinit commands, before any "built in" command is sent, in usecs (1million usec= 1sec)
at_after_preinit_pause=500000
;custom commands to be sent after the "built in" initialization commands, and what the modem is expected to send as reply
;the first empty string stop the postinit sending
;at_postinit_1=atcucu ; nonsense entry, just to show the postinit
at_postinit_1=at+cmic=0,9 ; modem's microphone sensitivity (our spk)
at_postinit_1_expect=OK
at_postinit_2=AT+CKPD="EEE" ;send three "end" buttonpress, to have the phone in a sane state, ready to dialing with furter CKPDs ***THIS IS IMPORTANT, needed on c650***
at_postinit_2_expect=OK
at_postinit_3=AT+CSSN=1,0
at_postinit_3_expect=OK
at_postinit_4=at+sidet=0 ; no sidetone in modem, please
at_postinit_4_expect=OK
at_postinit_5=at+clvl=99 ; modem's speaker level, out mic
at_postinit_5_expect=OK
;what command to query the battery status, and what the modem is expected to send as reply
at_query_battchg=AT+CBC
at_query_battchg_expect=OK
;what command to query the signal status, and what the modem is expected to send as reply
at_query_signal=AT+CSQ
at_query_signal_expect=OK
;what command to send a DTMF
at_send_dtmf=AT+VTS
;the modem will send us the following messages to signal that the visual indicators on the phone has changed because of events (without us to ask for them), loosely based on ETSI standard (see CIND/CIEV/CMER in ETSI). Variable by manufacturer and phone model
; no service
at_indicator_noservice_string=+CIEV: 2,0
; no signal
at_indicator_nosignal_string=+CIEV: 5,0
; low signal
at_indicator_lowsignal_string=+CIEV: 5,1
; low battery
at_indicator_lowbattchg_string=+CIEV: 0,1
; no battery battery
at_indicator_nobattchg_string=+CIEV: 0,0
; call is up
at_indicator_callactive_string=+CIEV: 3,1
; call is down
at_indicator_nocallactive_string=+CIEV: 3,0
; call is no more in process
at_indicator_nocallsetup_string=+CIEV: 6,0
; call incoming is in process
at_indicator_callsetupincoming_string=+CIEV: 6,1
; call outgoing is in process
at_indicator_callsetupoutgoing_string=+CIEV: 6,2
; remote party is ringing because of our call outgoing
at_indicator_callsetupremoteringing_string=+CIEV: 6,3
;call processing unsolicited messages, proprietary for each phone manufacturer
;the modem will send us the following mesage to signal that the line is idle (eg. after an outgoing call has failed, or after hangup)
at_call_idle=+MCST: 1
;the modem will send us the following mesage to signal that there is an incoming voice call
at_call_incoming=+MCST: 2
;the modem will send us the following mesage to signal that there is an active call (eg. the remote party has answered us, or we answered them)
;at_call_active=+MCST: 3
at_call_active=+CSSI: 7
;the modem will send us the following mesage to signal that our outgoing call has failed
at_call_failed=+MCST: 65
;the modem will send us the following mesage to signal that our outgoing call is in the calling phase
;at_call_calling=+MCST: 64
at_call_calling=+CSSI: 1
please copy here the debug
please copy here the debug output you get after:
set debug 100
Ok, so this was weird. It
Ok, so this was weird. It restarted Asterisk due to an error after the call was started:
trixbox1*CLI> set debug 100
Core debug is at least 100
-- Registered SIP '200' at 10.0.0.249 port 50754 expires 3600
-- Saved useragent "X-Lite release 1103k stamp 53621" for peer 200
Really destroying SIP dialog '75b9f3082ed228dc79cf27c8163f9bc7@10.0.0.254' Method: OPTIONS
Really destroying SIP dialog '1e48c421152cf3082510066165d299cd@10.0.0.254' Method: NOTIFY
-- Executing [18157185089@from-internal:1] Macro("SIP/200-09d5b710", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/200-09d5b710", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/200-09d5b710", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/200-09d5b710", "1|Set|REALCALLERIDNUM=200") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/200-09d5b710", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/200-09d5b710", "AMPUSERCIDNAME=TechTestExt") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/200-09d5b710", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/200-09d5b710", "AMPUSERCID=200") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/200-09d5b710", "CALLERID(all)="TechTestExt" <200>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/200-09d5b710", "REALCALLERIDNUM=200") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/200-09d5b710", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/200-09d5b710", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/200-09d5b710", "Using CallerID "TechTestExt" <200>") in new stack
-- Executing [18157185089@from-internal:2] Set("SIP/200-09d5b710", "_NODEST=") in new stack
-- Executing [18157185089@from-internal:3] Macro("SIP/200-09d5b710", "record-enable|200|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/200-09d5b710", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/200-09d5b710", "recordingcheck|20100117-170900|1263769740.0") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100117-170900|1263769740.0: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/200-09d5b710", "") in new stack
-- Executing [18157185089@from-internal:4] Macro("SIP/200-09d5b710", "dialout-trunk|2|18157185089||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/200-09d5b710", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/200-09d5b710", "0?sub-pincheck|s|1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/200-09d5b710", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/200-09d5b710", "DIAL_NUMBER=18157185089") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/200-09d5b710", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/200-09d5b710", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/200-09d5b710", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/200-09d5b710", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/200-09d5b710", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/200-09d5b710", "outbound-callerid|2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/200-09d5b710", "0|SetCallerPres|") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/200-09d5b710", "0|Set|REALCALLERIDNUM=200") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/200-09d5b710", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/200-09d5b710", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/200-09d5b710", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/200-09d5b710", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/200-09d5b710", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/200-09d5b710", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/200-09d5b710", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/200-09d5b710", "0|SetCallerPres|prohib_passed_screen") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/200-09d5b710", "1|AGI|fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
> fixlocalprefix: Using pattern 1+NXXNXXXXXX
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/200-09d5b710", "OUTNUM=18157185089") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/200-09d5b710", "custom=AMP") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/200-09d5b710", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/200-09d5b710", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/200-09d5b710", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/200-09d5b710", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/200-09d5b710", "1?customtrunk") in new stack
-- Goto (macro-dialout-trunk,s,21)
-- Executing [s@macro-dialout-trunk:21] Set("SIP/200-09d5b710", "pre_num=AMP:Celliax/line0/") in new stack
-- Executing [s@macro-dialout-trunk:22] Set("SIP/200-09d5b710", "the_num=OUTNUM") in new stack
-- Executing [s@macro-dialout-trunk:23] Set("SIP/200-09d5b710", "post_num=") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/200-09d5b710", "1?outnum:skipoutnum") in new stack
-- Goto (macro-dialout-trunk,s,25)
-- Executing [s@macro-dialout-trunk:25] Set("SIP/200-09d5b710", "the_num=18157185089") in new stack
-- Executing [s@macro-dialout-trunk:26] Dial("SIP/200-09d5b710", "Celliax/line0/18157185089|300|") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:27] Goto("SIP/200-09d5b710", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/200-09d5b710", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/200-09d5b710", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 42) - failing through to other trunks") in new stack
-- Executing [18157185089@from-internal:5] Macro("SIP/200-09d5b710", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/200-09d5b710", "all-circuits-busy-now|noanswer") in new stack
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing [s@macro-outisbusy:2] Playback("SIP/200-09d5b710", "pls-try-call-later|noanswer") in new stack
-- Playing 'pls-try-call-later' (language 'en')
-- Executing [s@macro-outisbusy:3] Macro("SIP/200-09d5b710", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/200-09d5b710", "vw") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/200-09d5b710", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/200-09d5b710", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/200-09d5b710", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/200-09d5b710", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/200-09d5b710", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/200-09d5b710'
trixbox1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
[trixbox1.localdomain asterisk]# /usr/sbin/safe_asterisk: line 125: 12493 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} < /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
[trixbox1.localdomain asterisk]#
how to obtain debug infos
after a trixbox restart
you probably have to change the permissions again on ALL:
chown -r asterisk:asterisk
chown -r asterisk:asterisk /dev/ttyA*
chown -r asterisk:asterisk /dev/ttyU*
chown -r asterisk:asterisk /dev/snd
(you may want to put those commands in the script that launches trixbox, but we'll leave that for after)
then follow the instruction on http://www.gsmopen.org/howtobug
on how to edit the logger.conf file of asterisk to enable debugging
then from the asterisk command line:
set debug 100
load chan_celliax.so
then retry the call and report here the debugging
(you have to enable the debugging, what you posted is without debugging)
Sorry, should have known thats what you meant.
Here are the errors in the debug output:
[Jan 17 17:28:43] ERROR[12784]: celliax_additional.c:147 alsa_open_dev: rev exported[(nil)|b7882b90][ERROR 147 ][line0 ][-1, 0, 0] snd_pcm_open failed with error 'No such file or directory' on device 'plughw:1', if you are using a plughw:n device please change it to be a default:n device (so to allow it to be shared with other concurrent programs), or maybe you are using an ALSA voicemodem and slmodemd is running?
[Jan 17 17:28:43] ERROR[12784]: celliax_additional.c:37 alsa_init: rev exported[(nil)|b7882b90][ERROR 37 ][line0 ][-1, 0, 0] Failed opening ALSA capture device: plughw:1
[Jan 17 17:28:43] ERROR[12784]: chan_celliax.c:1155 celliax_new: rev exported[(nil)|b7882b90][ERROR 1155 ][line0 ][-1, 0, 0] Failed initializing sound device
Looks like I do not have the soundcard setup properly. (FYI-There are no soundcards in the machine, just the one builtin to the mobigater)
Thanks!
wrong audio device
redo all the permission changing, then:
as in that page of how to report bugs, can you please paste here the results of:
aplay -l
(you probably have to modify the celliax.conf file, and change plughw:1 with plughw:0 in all instances)
You were very right!!! It
You were very right!!!
It completed the call! No audio yet though. But let me send an incoming call into something that will give some audio.
Just a min....
:) Im so exicted! :) - Maybe I will get this project to my boss afterall!
THANK YOU! I will get back to you on what I find.
hey stop!
stop a moment!
:)
you have to check the volumes with this linux (not asterisk) command:
alsamixer -Vall
alsamixer -Vcapt
press the spacebar on mic when you have done the -Vcapt command
then you'll have audio, probably
you need to activate capture on mic
press the spacebar on mic when you have done the -Vcapt command
so it will appear the "CAPTURE" in red
Ok, let me do that.
Ok, let me do that.
Looks one way audio. I can
Looks one way audio. I can hear what is spoken into the SIP phone on my cell phone, but not vice versa.
Any thoughts?
raise the speaker volume
with:
alsamixer -Vplay
raise the speaker volume
Also - all calls IN to the
Also - all calls IN to the trunk just plays the vm-goodbye audio, and hangs up.
vm-goodbye
you have to set in the celliax.conf file the context and extension of destination
copy it from the other configuration files of asterisk (I don't know which trixbox wants)
Thank you! I will work with
Thank you!
I will work with this and give it a try. I will post a detailed howto for asterisk after I have everything figured out so that others dont have to spend as much time as I did figuring out how to install the channel driver.
Thank you again.
thank you to you!
And please remember to post here the HOWTO
I will transform it in a page!
-giovanni
also, raise the speaker volume
read http://www.gsmopen.org/node/51#comment-113
I definatly will! Also, why
I definatly will!
Also, why would this not work?
; Default context (is overridden with @context syntax)
;
context=default
;
; Default extension (in extensions.conf) where incoming calls land
;
extension=s
wouldnt that drop it into asterisks incoming routes?
trixbox has its own defaults
trixbox has its own default incoming destination, if I remember correctly
the one you wrote (from the celliax.conf) would work with plain asterisk, trixbox modifies the defaults of asterisk, trixbox modifies the config files of asterisk with its own defaults
Asterisk Fails to load on bootup
Everything seems to be working well. Only problem is that when asterisk loads on a bootup, it fails because it does not have correct permissions to the /dev/ttyACM0 . However if I restart asterisk, it runs fine. I put commands in the rc.5 that modify the permissions, is there a chance that asterisk is trying to start before this level?
bootup, and glad to know you succeeded
mmmh, it will probably start in rc.3
Also, if I remember correctly, trixbox has its own start script, and/or a safe_asterisk script
look into /etc/init.d/*
Try to poke around to find where they are and put the commands there (BOTH into the start trixbox script AND into the safe_asterisk script).
(also if they are executed more than one time, no problem).
Btw, you're very fast - if you're really new to asterisk and linux ;)
Thanks again, I will give
Thanks again, I will give that a whirl!
Also, the pbx still does not register any DTMF keys pressed on the calling phone, and does not hear anything said into the calling phone.
I have the mic turned up all the way. Any suggestions?
Thanks for the comments, makes me feel much better about myself!
Also, I was putting the chown
Also, I was putting the chown command in the rc.local, when does that run?
Figured out the Oneway audio
Figured out the Oneway audio issue. I wasnt pressing 'spacebar' on the mic capture settings. If i just read the directions properly, lol.
Also will these settings keep on a reboot?
rc.local last
rc.local normally runs as last, but maybe trixbox-asterisk are started before...
keeping alsa (soundcard) settings on reboot
after you set it to your like, execute the command (from linux):
alsactl store
then, edit rc.loacl and put there:
alsactl restore
this way it will restore the settings you "photographed" with alsact store at each reboot
Audio Issues.
Hi Everyone! I've successfully configured celliax, asterisk to work with my Nokia E90. I'm using only a USB cable. I need to transfer audio. Any ideas?
Audio issues 2
You need an audio cable to bring the audio from the handsfree jack to the computer soundcard.
You need to build that cable, or have someone build it for you. The speaker of the handsfree jack goes to the microphone of the soundcard, and the microphone of the handsfree jack goes to the speaker of the soundcard.
You can start with an handsfree for your cellphone, and an headset for your computer.
You cut the cable of both, and you join the cables' parts with the connector crossed.
You end up havint one only cable, with a jack for the handsfree at one end, and the two jacks for mic and speak of the soundcard at the other end (and some junk you trow away :) ).
Audio issues 3
Thanks a lot. I will try that out now
Audio Issues 4
Thanks a lot guys. I made the audio cable and it is working now. There is another issue:
Asterisk answers an incoming call on that channel,
Plays a welcome message, and then
dials a sip phone.
Once the connection is established, audio doesnt get transmitted.
Any ideas?
audio issues 5
please check you got the volumes of playback and capture correct:
http://www.gsmopen.org/alsaconf
if the problem persists, please follow the guidelines to report bugs, so we can be of help:
http://www.gsmopen.org/howtobug
then open a new topic with a descriptive title, so it can be useful to other too
ciao for now! :)
audio issues 6
Thanks. I fixed the volume issues.
Next Question:
How does celliax handle call waiting?
Thanks
Call waiting (no such thing)
Maybe I don't understand the question.
If you mean the call waiting function that the carrier give you if you activate it in the SIM, there is no particular handling of call waiting, disable it in the SIM :).
Call waiting
That means, i can only receive one call at a time on that line? How about call queueing?
No call queueing, but please describe it for future
Yes, definitely yes. One SIM one call.
But, for a future release "feature request", how do you would like the call waiting/call queueing feature to behave?
I mean, let's say you have activated call waiting with your carrier, and let's say when the second (or third, etc) incoming call arrives is signaled by the AT commands (not sure about that, but let's suppose).
So? What would happen at the level of the interface? How would the interface deal with that?
I mean, one SIM can have only one call active at time... So?
Please, describe the desired behavior so maybe can be added to a future release
No call queueing 2
Best would be if you have in mind a product that do this, and post here a link to the technical documentation of it.
Call queuing/waiting
In this case, we're dealing with call waiting. That means that there is an Asterisk channel already up and connected a celliax channel associated with the physical interface. Now, we have another call coming in destined for this interface. Can we go ahead and set up this call with a psuedo channel (or something like that)? Then have asterisk playback a waiting tone to the caller(or go through a dialplan?)
Thanks
call queueing/waiting 2
Mmmmmh, in any case you can deal with just one call at time, putting them on hold and off hold at turn. The dialplan or whatever would be extremely complex and error prone....
I'll think at that, but I don't see it really feasible...
Hello Chuck - re. your MobiGater
Hi Chuck,
as Giovanni advised me to buy MobiGater to work for Asterisk for Windows
I would really appreciate your help
and assistance in setting up celliax.conf to have MobiGater to work for me.
I considered to use 3G usb voice modem as GSM Gateway.
The issue is 3G usb voice modem installs on 3 serial ports.
One for data (Internet), another for voice + SMS and one another for signalling (?)
I don't know what is an analogy between 3G usb voice modem and MobiGater,
as I don't know at commands set to work with a modem
to let it me to configure it as GSM Gateway.
I may ask Huawei about its configuration to have it to work for Asterisk for Linux / Windows
as GSM Gateway,
but still need one other man assistance and interest in such solution,
as I am an interaction man by nature.
Please PM me if you are interested in such solution.
dariusjack2006@yahoo.ie
E169, E160 are both 3G usb voice modems avaliable for tests.
I have yet to get voice enabled in E160.
Anyone been able to get the
Anyone been able to get the kinks out yet?
Tom - Sniper Rifle