having problems with x-lite

davies
Posts: 16
Member Since:
2008-06-30

after my x-lite softphone is launched on linux. it shows logged in but
latter displays "unable to open audio: no audio drive". at this time i
will receive no audio.

if my extension is called , when there is audio, the caller will here it
ring on his own headset but i will not hear it ring. x-lite would only
notify me that there is an incomming call on line 3.

when the call is picked, we will not here ourselves. especially when
calling between 2 linux systems or when using usb phones.

if call is between a linux to windows system , the call goes through
but in the reverse case one of the ends won't hear the other.

i configured the x-lite phone by adding only the user extension and the
secret number and the sip proxy as the IP address of my trixbox server.
please kindly help me solve this problem as soon as you can. thank you



SkykingOH
Posts: 9680
Member Since:
2007-12-17
Fix your linux audio

Fix your linux audio drivers.

--

Scott

aka "Skyking"



atilio
Posts: 288
Member Since:
2006-06-01
You can use kmix or aumix to

You can use kmix or aumix to adjust mic and headset volumes. Sometimes the default will be muted. We use the Logitechs USB 250 in our call center and they are both auto detected on both Linux (Centos 5) and Windows (XP).



davies
Posts: 16
Member Since:
2008-06-30
still having problems with x-lite

thank you for your reply, but i have tried it and x-lite is still sending the same no audio device message. i tested my clients audio by played audio files on it and it walked fine. but while using the x-lite it stil siezes audio. and the best we can have is one-way - audio.
using (service iptable status) i tested the state of the fire wall by performing ssh into the server. the result was that it was stopped. i then tried starting it but it didn't start. then i checked if the udp port 5060 is running on it by using (telnet 192.168.19.6 5060) but it showed connection refused.
i also tried the rtp port , it was the same result.but i latter tried the http 80 and it showed connected. but the other were not connected. please note that my major concern for now is to make calls within my local network. this is my sip.conf /etc/asterisk/sip.conf (trixbox v2.4)

[general]
;
; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=yes

; These will all be included in the [general] context
;
#include sip_general_additional.conf
#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf
#include sip_custom_post.conf
THIS WAS ALL THE FILE HAD , I THEN ADDED THE FOLLOWING FOR THREE OTHER EXTENTIONS

sip_additional.conf

[2000]
type=friend
secret=2000
record Incoming=On Demand
record Outgoing=On Demand
qualify=yes
port=5060
pickupgroup=
nat=no
mailbox=2000@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/2000
context=from-internal
canreinvite=no
callgroup=
callerid=device
accountcode=
call-limit=50

[2001]

AND THIS IS MY extensions.conf /etc/asterisk/extensions.conf PLEASE NOTE THAT THIS IS JUST THE LAST POTION OF THE FILE IT IS A VERY VOLUMINOUS FILE AND I HAVEN'T ADDED ANY LINE TO IT.

; ############################################################################
; Extension Contexts [ext]
; ############################################################################

[ext-fax]
exten => s,1,Answer
exten => s,2,Goto(in_fax,1)
exten => in_fax,1,StopPlayTones
exten => in_fax,2,GotoIf($["${FAX_RX}" = "system"]?3:analog_fax,1)
exten => in_fax,3,Macro(faxreceive)
exten => in_fax,4,Hangup
exten => analog_fax,1,GotoIf($["${FAX_RX}" = "disabled"]?4:2) ;if fax is disabled, just hang up
exten => analog_fax,2,Set(DIAL=${DB(DEVICE/${FAX_RX}/dial)});
exten => analog_fax,3,Dial(${DIAL},20,d)
exten => analog_fax,4,Hangup
;exten => out_fax,1,wait(7)
exten => out_fax,1,txfax(${TXFAX_NAME},caller)
exten => out_fax,2,Hangup
exten => h,1,system(/var/lib/asterisk/bin/fax-process.pl --to ${EMAILADDR} --from ${FAX_RX_FROM} --subject "Fax from ${URIENCODE(${CALLERID(number)})} ${URIENCODE(${CALLERID(name)})}" --attachment fax_${URIENCODE(${CALLERID(number)})}.pdf --type application/pdf --file ${FAXFILE});
;this is where parked calls go if they time-out. Should probably re-ring
[default]
include => ext-local
exten => s,1,Playback(vm-goodbye)
exten => s,2,Macro(hangupcall)

please help me as soon as you can.



Comment viewing options

Select your preferred way to display the comments and click "Save settings" to activate your changes.