system toolbox » voip

uspio sam sinoć na nmraka-5 buidati freeswitch, te osnovne stvari pohvatati što se tiče konfiguracije

ono što me je posebno zainteresovalo jeste podstavka mod_skypiax i to sam na kraju uspio

istina, nakon par poziva, freeswitch ode cpu na 50% i ne može se uraditi shutdown bez kill-a

mežutim kvaliteta zvuka je dobra i stvar obećavajuće izgleda općenito

hernad@nmraka-5:~/devel/git/freeswitch/src/mod/endpoints/mod_skypiax/configs$ sudo bash startskype_hernad.sh

# remember to add here the removing of all the installed snd-* modules, so you're sure only the snd-dummy driver willbe around
rmmod snd_hda_intel

# you need three dummy soundcard for 20 Skype client instances, because each dummy soundcard can handle a max of 8 Skype instances
# the enable= module parameter tells how many cards to start. For each additional card, add a comma and a 1
# manually configure the first 8 Skype client instances to use the hw:Dummy_0, the next 8 instances to use hw:Dummy_1, etc for all three devices (Play, Capture, Ring)
modprobe snd-dummy enable=1,1,1
sleep 3

#start the fake X server on a given port
/usr/bin/Xvfb :101 -auth /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf &
sleep 3

# as user "unixusername", start a Skype client instance that will connect to the X server above, and will login to the Skype network using the 'username password' you send to it on stdin. Here xxx would be the Skype password and skypiax1 the Skype username. Please note that the ".Skypiax" directory containing the Skype clients configurations MUST be in the home directory of the "unixusername" user.
su hernad -c "/bin/echo 'hernad moja_skype_passwd'| DISPLAY=:101 /usr/bin/skype --pipelogin &" 

prije toga sam napravio skypiax_auth i prema README-u ovom:

hernad@nmraka-5:~/devel/git/freeswitch/src/mod/endpoints/mod_skypiax/configs$ ls README.skypiax_auth

uradio podešenja klijenta

hernad@nmraka-5:~/devel/git/freeswitch/src/mod/endpoints/mod_skypiax/configs$ ps ax | grep Xv

 9594 pts/4    S      0:00 /usr/bin/Xvfb :101 -auth /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf

hernad@nmraka-5:~/devel/git/freeswitch/src/mod/endpoints/mod_skypiax/configs$ ps ax | grep skype

 9607 pts/4    Sl     0:05 /usr/bin/skype --pipelogin

u freeswitch konzoli:

freeswitch@nmraka-5> load mod_skypiax

2009-04-23 12:38:23 [WARNING] mod_skypiax.c:950 load_config() rev exported[(nil)|37     ][WARNINGA  950  ][hernad    ][-1, 0, 0] STARTING interface_id=1
2009-04-23 12:38:23 [NOTICE] mod_skypiax.c:976 load_config() rev exported[(nil)|37     ][NOTICA  976  ][hernad    ][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1
2009-04-23 12:38:23 [NOTICE] mod_skypiax.c:986 load_config() rev exported[(nil)|37     ][NOTICA  986  ][hernad    ][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=1, waiting 60 seconds for CURRENTUSERHANDLE==hernad
API CALL [load(mod_skypiax)] output:
+OK

2009-04-23 12:38:23 [WARNING] mod_skypiax.c:1004 load_config() rev exported[(nil)|37     ][WARNINGA  1004 ][hernad][-1, 0, 0] Interface_id=1 is now STARTED, the Skype client to which we are connected gave us the correct CURRENTUSERHANDLE (hernad)
2009-04-23 12:38:23 [CONSOLE] switch_loadable_module.c:889 switch_loadable_module_load_file() Successfully Loaded [mod_skypiax]
2009-04-23 12:38:23 [NOTICE] switch_loadable_module.c:142 switch_loadable_module_process() Adding Endpoint 'skypiax'
2009-04-23 12:38:23 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'sk'
2009-04-23 12:38:23 [NOTICE] switch_loadable_module.c:270 switch_loadable_module_process() Adding API Function 'skypiax'

hernad@ubuntu:~/devel/freeswitch$ make -C src/mod/endpoints/mod_skypiax/

make: Entering directory `/home/hernad/devel/freeswitch/src/mod/endpoints/mod_skypiax'
make[1]: Entering directory `/home/hernad/devel/freeswitch/src/mod/endpoints/mod_skypiax'
Compiling skypiax_protocol.c...
Compiling mod_skypiax.c...
quiet_libtool: compile:  gcc -DSKYPIAX_SVN_VERSION=\"exported\" -I/home/hernad/devel/freeswitch/src/include -I/home/hernad/devel/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_skypiax.c  -fPIC -DPIC -o .libs/mod_skypiax.o
quiet_libtool: compile:  gcc -DSKYPIAX_SVN_VERSION=\"exported\" -I/home/hernad/devel/freeswitch/src/include -I/home/hernad/devel/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -DHAVE_CONFIG_H -c mod_skypiax.c -o mod_skypiax.o >/dev/null 2>&1
Creating mod_skypiax.la...
make[1]: Leaving directory `/home/hernad/devel/freeswitch/src/mod/endpoints/mod_skypiax'
make: Leaving directory `/home/hernad/devel/freeswitch/src/mod/endpoints/mod_skypiax'

hernad@ubuntu:~/devel/freeswitch$ sudo make -C src/mod/endpoints/mod_skypiax/ install

make: Entering directory `/home/hernad/devel/freeswitch/src/mod/endpoints/mod_skypiax'
make[1]: Entering directory `/home/hernad/devel/freeswitch/src/mod/endpoints/mod_skypiax'
installing mod_skypiax.la
quiet_libtool: install: warning: relinking `mod_skypiax.la'
make[1]: Leaving directory `/home/hernad/devel/freeswitch/src/mod/endpoints/mod_skypiax'
make: Leaving directory `/home/hernad/devel/freeswitch/src/mod/endpoints/mod_skypiax'

http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk

hm ali ovdje imam problem jer ja hoću interno da se spojim

  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>

  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/>

stavio internal_auth_calls=false

nakon toga pokušaj ifold se registruje barem:

freeswitch@ubuntu> 2009-04-23 16:54:24 [WARNING] sofia.c:4216 sofia_handle_sip_i_invite() IP 192.168.45.4 Rejected by acl "domains" 
2009-04-23 16:54:34 [WARNING] sofia.c:4216 sofia_handle_sip_i_invite() IP 192.168.45.4 Rejected by acl "domains" 

petljao svašta nešto, glavni problem je bio što asterisk tretira vanjske sip i rtp ip adrese isključivo internet adrese, pa pokušava prići svana, zato sam ja sada osakatio to i stavio da su sip i rtp external adrese = local_ip_v4 adresa

u ivr meniju podesio:

root@ubuntu:/usr/local/freeswitch# cat conf/autoload_configs/ivr.conf.xml | grep bridge

      <entry action="menu-exec-app" digits="1" param="bridge sofia/external/061141311@192.168.45.4:5060"/>

tako da pritiskom na 1 pozivamo moj mobitel

takođe me je nakon ovih promjena postavki prebacio sa default.xml public.xml plan

root@ubuntu:/home/hernad# sudo apt-get install xvfb

na 192.168.45.143 imam problem što skype ne prepoznaje dummy modul

imam dumy, ali mi se ne pojavljuje hw:dummy kao opcija u skype-u

hernad@ubuntu:~$ asoundconf list

Names of available sound cards:
Solo1
default
Dummy
Dummy_1
Dummy_2

evo šta na mom notebook-u na kome sam skype podesio bez probelema imam:

hernad@nmraka-5:~/devel/git$ asoundconf list

Names of available sound cards:
SB
HDMI
HDMI_1
Dummy
Dummy_1
Dummy_2

ne kontam

podesio sam ivr demo tako da kada ućem u ovaj meni (ekstenzija 5000 u dialplan-u) 520 je recimo poziv bjasko mobitel, a kasnije sam skontao kako da ukucam čitav 033xxxxxx broj - pogledaj conf

root@ubuntu:~# cat /usr/local/freeswitch/conf/autoload_configs/ivr.conf.xml

<configuration name="ivr.conf" description="IVR menus">                              
  <menus>                                                                            
    <!-- demo IVR setup -->                                                          
    <!-- demo IVR, Main Menu -->                                                     
    <menu name="demo_ivr"                                                            
          greet-long="phrase:demo_ivr_main_menu"                                     
          greet-short="phrase:demo_ivr_main_menu_short"                              
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"                      
          exit-sound="voicemail/vm-goodbye.wav"                                      
          timeout ="90000"                                                           
          inter-digit-timeout="3000"                                                 
          max-failures="3"                                                           
          max-timeouts="3"                                                           
          digit-len="10">            <<<<<<<<<<<<<-------------                                                
      <entry action="menu-exec-app" digits="1" param="bridge sofia/external/061141311@192.168.45.4:5060"/>   <<<<<<<-----------   
      <entry action="menu-exec-app" digits="20" param="bridge sofia/external/033269290@192.168.45.4:5060"/>  <<<<<<<-----------
      <entry action="menu-exec-app" digits="500" param="bridge sofia/external/061141311@192.168.45.4:5060"/> <<<<<<<----------
      <entry action="menu-exec-app" digits="520" param="bridge sofia/external/061477105@192.168.45.4:5060"/>     <<<<<<<<<<<------------------                                                                  
      <entry action="menu-exec-app" digits="/^(033[0-9][0-9][0-9][0-9][0-9][0-9])$/" param="bridge sofia/external/$1@192.168.45.4:5060"/>             <<<<<<<<<<<<<-------------------------------------------------------
      <entry action="menu-exec-app" digits="2" param="transfer 9996 XML default"/>    <!-- FS echo -->
      <entry action="menu-exec-app" digits="3" param="transfer 9999 XML default"/>    <!-- MOH -->
      <entry action="menu-exec-app" digits="4" param="transfer 9991 XML default"/>    <!-- ClueCon -->
      <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>       <!-- Screaming monkeys -->
      <entry action="menu-sub" digits="6" param="demo_ivr_submenu"/>                  <!-- demo sub menu -->
      <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
      <entry action="menu-top" digits="9"/>          <!-- Repeat this menu -->
    </menu>
    <!-- Demo IVR, Sub Menu -->
    <menu name="demo_ivr_submenu" 
        greet-long="phrase:demo_ivr_sub_menu" 
        greet-short="phrase:demo_ivr_sub_menu_short" 
        invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" 
        exit-sound="voicemail/vm-goodbye.wav" 
        timeout="15000" 
        max-failures="3" 
        max-timeouts="3">
      <entry action="menu-top" digits="*"/>
     </menu>

  </menus>

</configuration>

<!-- TTS sample; non-functional but it demonstrates say: and TTS -->
<!--
    <menu name="demo3" 
          greet-long="say:Press 1 to join the conference, Press 2 to join the other conference" 
          greet-short="say:Press 1 to join the conference, Press 2 to join the other conference" 
          invalid-sound="say:invalid extension" 
          exit-sound="say:exit sound" 
          timeout ="15000" 
          max-failures="3">
      <entry action="menu-exit" digits="*"/>
      <entry action="menu-playback" digits="1" param="say:You pressed 1"/>
      <entry action="menu-call-transfer" digits="2" param="1000"/>
      <entry action="menu-call-transfer" digits="3" param="1001"/>
    </menu>
-->

inače napravljen je sip bridge između fs-a i ifold-a

asterisk sip.conf

[freeswitch]
type=friend
username=freeswitch
password=sar........
host=dynamic
callerid="freeswitch" 
callgroup=1
qualify=yes
dtmfmode=rfc2833
nat=no
canreinvite=yes