[Freeswitch-users] problems with outgoing calls

Czaderna pawel.czaderna at mainseek.com
Tue May 13 04:49:18 PDT 2008


Hello :)

At the begining - sorry for my english.
I have following problems with freeswitch:

1. I have got 3 computers in LAN. On one computer is running freeswitch, and on the others are intalleted Xlite software (sip clients).

2. On "voipdiscount.com" I've got an account for free local calls.

3. I want make a call on mobile phone using freeswitch and voipdiscount.com So on my computer I start XLite, log in on default freeswitch account 1000 and then try dial a mobile phone number

4. On dialplan I have something like this:

./CONF//DIALPLAN: public.xml
<include>
  <context name="public">

    <extension name="public_extensions">
     <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

    <extension name="test_voipdiscount">
        <condition field="destination_number" expression="^([0-9]{13})$">
            <action application="bridge" data="sofia/gateway/voipdiscount.com/$1 at sip.voipdiscount.com"/>
        </condition>
    </extension>

  </context>
</include>

and on the sip_profiles (conf/sip_profiles/outbound/voipdiscount.com.xml) I have following configuration of gateway:

<include>
  <gateway name="voipdiscount.com">
  <param name="username" value="user_name"/>
  <param name="realm" value="sip.voipdiscount.com"/>
  <param name="password" value="user_password"/>-->
  <param name="proxy" value="sip.voipdiscount.com"/>
  </gateway>
</include>


After connection on freeswitch I get following messages:


2008-05-13 09:51:43 [INFO] mod_iax.c:928 mod_iax_runtime() IAX Ready Port 4569
freeswitch at xx> 2008-05-13 09:51:43 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering voipdiscount.com
2008-05-13 09:51:43 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering ideasip.com
2008-05-13 09:51:47 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1000 at 192.168.3.79 [701574c2-20c1-11dd-9725-39613e150eca]
2008-05-13 09:51:47 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1000->0048606xxxxxx at public
2008-05-13 09:51:47 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/outbound/0048606xxxxxx at sip.voipdiscount.com [70165892-20c1-11dd-9725-39613e150eca]
2008-05-13 09:52:14 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering voipdiscount.com
2008-05-13 09:52:14 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering ideasip.com
2008-05-13 09:52:46 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering voipdiscount.com
2008-05-13 09:52:46 [NOTICE] sofia_reg.c:104 sofia_reg_check_gateway() registering ideasip.com
2008-05-13 09:52:47 [NOTICE] switch_ivr_originate.c:1154 switch_ivr_originate() Hangup sofia/outbound/NTK at sip.voipdiscount.com [CS_HOLD] [NO_ANSWER]
2008-05-13 09:52:47 [INFO] mod_dptools.c:1551 audio_bridge_function() Originate Failed.  Cause: ORIGINATOR_CANCEL
2008-05-13 09:52:47 [NOTICE] mod_dptools.c:1578 audio_bridge_function() Hangup sofia/default/1000 at 192.168.3.79  [CS_EXECUTE] [ORIGINATOR_CANCEL]
2008-05-13 09:52:47 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 1 (sofia/default/1000 at 192.168.3.79 ) Ended
2008-05-13 09:52:47 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1000 at 192.168.3.79 [CS_HANGUP]
2008-05-13 09:52:47 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 2 (sofia/outbound/0048606xxxxxx at sip.voipdiscount.com) Ended
2008-05-13 09:52:47 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/outbound/0048606xxxxxx at sip.voipdiscount.com [CS_HANGUP]


There are no errors but after few minutes connection is hang up (no calling signal on the channnel).

I don't know where is the problem.
Also I was trying connect through ideasip.com (got SIP number and an account at this provider) same as above, but in the dialplan instead of "bridge" I used "redirect". So my dialplan looks like this:

<include>
  <context name="public">

    <extension name="public_extensions">
     <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

    <extension name="test_voipdiscount">
        <condition field="destination_number" expression="^([0-9]{11})$">
     <!-- It works with the following line -->
            <action application="redirect" data="sip:$1 at proxy.ideasip.com"/>
     <!-- below line doesn't work -->
     <!-- <action application="bridge" data="sofia/gateway/ideasip.com/$1 at sip.ideasip.com"/> -->
        </condition>
    </extension>

  </context>
</include>

On voipdiscount.com this trick doesn't working.

Could anyone help me, please ?

Thanks
Paul.
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