how to answer to incoming call from cellphone

hi to everyone.
i've installed celliax on trix box 2.2.5
when i make a call from internal trought celliax works! but if a call the cellphone it answer but the internal don't ring.
an other question from operator pannel isnt the celliax trunk! why?

any suggestion?

P.S. i use a motorola 650 and this is my celliax.conf

Celliax Asterisk Driver
;
; Configuration file
; lines beginning with semicolon (" are ignored (commented out)
;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;
; The first interface (named nicephone)
[nicephone]
;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; general settings, valid on all platforms
;
;
; Default language
;
language=en
;
; Default context (is overridden with @context syntax)
;
context=[mobile]
;
; 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_skype=yes
debug_call=yes
;debug_cvm=yes

; wanna skype support?
skype=yes

;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; 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 Motorola phone, recognized as a modem by Linux
;control_device_name=/dev/ttyS9 ; this is the COM10 on windows
;control_device_name=/dev/ttyUSB0 ; this is the first USB serial port in Linux

;cvm_subscription_1_pin = 0000 ; CVM hold registration info about two FP (DECT base stations)
;cvm_subscription_2_pin = 1234 ; these are PINs for each FP, used only during registartion process

;cvm_subscription_no = 1 ; CVM can be locked to one FP at the time, please choose 1 or 2

;cvm_serial_delay = 1 ; used to fix problems with serial communication, probably not needed any more, please see alsa settings, it looks for me that I have a problem with CM108 and PL2303HX on USB.
;cvm_volume_level = 1 ; volume level control in CVM (0-9)

;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 ; for usb audio cm-108 on motorola c650
capture_boost=0 ; for usb audio cm-108 on motorola c650
;playback_boost=10 ; for integrated mobo audio for motorola c650
;capture_boost=10 ; for integrated mobo audio for motorola c650

;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; alsa settings, ignored in windows
;
;alsa internal settings, if in doubt, leave them alone ;-)
;to check and do the audio fine tuning, use microphone and headset directly connected to the soundcard, is easier and cheaper
;period_size, if you hear a scratchy background noise, increase it to 170 and check if is better
;CVM - these alsa setting give good sound quality, CM108 USB card + PL2303HX Serial converter
;alsa_period_size=32
;alsa_periods_in_buffer, if you get too many XRUNs, or if you get noise after EAGAINs, increase it (it will increase your alsa latency, but is nothing. The real latency is introduced by the round trip of audio from cellphone to cellphone)
;alsa_periods_in_buffer=32

alsa_period_size=160
alsa_periods_in_buffer=4

;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=default:0
;alsa_playback_device_name=default:0

;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; window multimedia settings, ignored in linux
;
;names of the sound devices in windows, this is the first soundcard, the second would be 1
;winmm_capture_device_name=0
;winmm_playback_device_name=0

;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;
; 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
;what the modem is expecting in the part of the dial command after the number to be dialed (eg: ;)
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"
;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=1000000
;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=1000000
;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_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_postinit_3_expect=
at_postinit_4=
at_postinit_4_expect=
at_postinit_5=
at_postinit_5_expect=

;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

;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
;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

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;
;;;another interface, called nokianice, commented out. Remove two semicolon (" from the beginning of all the following lines to activate it
;;[nokianice]
;;
;;; skype is not yet supported, but will come back in the future
;;skype=no
;;
;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;; serial settings, valid for all platforms
;;;
;;; control_device_protocol can be AT or FBUS2 or NO_SERIAL
;;control_device_protocol=fbus2
;;
;;;speed of the serial port
;;control_device_speed=115200
;;
;;;name of the serial port device
;;control_device_name=/dev/ttyUSB0 ; this is the first USB serial port in linux
;;;control_device_name=/dev/ttyS11 ; this is COM12 in windows
;;
;;;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=1
;;
;;;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
;;;
;;;boost can be positive or negative (-40 to +40) in db
;;;experiment to find which values are best for your soundcard
;;playback_boost=10 ; for integrated mobo audio for nokia3110
;;capture_boost=-5 ; for integrated mobo audio for nokia3110
;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;; alsa settings, ignored in windows
;;;
;;;alsa internal settings, if in doubt, leave them alone ;-)
;;;to check and do the audio fine tuning, use microphone and headset directly connected to the soundcard, is easier and cheaper
;;;period_size, if you hear a scratchy background noise, increase it to 170 and check if is better
;;alsa_period_size=170
;;;alsa_periods_in_buffer, if you get too many XRUNs, or if you get noise after EAGAINs, increase it (it will increase your alsa latency, but is nothing. The real latency is introduced by the round trip of audio from cellphone to cellphone)
;;alsa_periods_in_buffer=4
;;
;;;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
;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;;;;;;;;;;;;;;;;;;;;;;;;;;
;;; window multimedia settings, ignored in linux
;;;
;;;names of the sound devices in windows, this is the second soundcard, the first would be 0
;;winmm_capture_device_name=1
;;winmm_playback_device_name=1
;;

;Configured for LINUX
control_device_protocol=AT
alsa_capture_device_name=default:0
alsa_playback_device_name=default:0
control_device_name=/dev/ttyACM0


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if it works like 2.3.x

For having the internal extension to ring when there is an incoming celliax call, you have to define an inbound route, please see the http://www.celliax.org/node/412 guide for an example.

If trix 2.2.5 works the same as 2.3.x, for having the celliax trunk to show in Flash Operator Panel you have to edit a file, /var/www/html/panel/op_buttons_custom.cfg and add the following lines:

[Celliax/nicephone]
Position=53
Label="Celliax nicephone"
Extension=-1
Icon=3
Panel_Context=default

Giovanni


internal don't ring

hi Givanni, i followed the instruction, but the internal don't ring. the cellphone answers tself after 1 ring.
where'e the mistake?

many thanks.


more info needed

Please post here the debugging output, or we have no mean to understand your problem


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