[Freeswitch-users] problems with outgoing calls
Brian West
brian at freeswitch.org
Tue May 13 14:27:31 PDT 2008
On May 13, 2008, at 6:49 AM, Czaderna wrote:
> 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>
>
tip.. I see your bridge line is a bit over worked. sofia/gateway/
voipdiscount.com/$1 is plenty then set from-domain in your gateway
properly.
> </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>
>
Why are you setting a realm? They should tell you the realm when they
challenge you... only rare cases should you ever need to fill out the
realm.
>
> 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).
Please try to collect the log with "console level debug" and see if
that gives you any more info. This looks like Originator Cancel was
the hangup cause but the debug might show a bit more info.
>
> 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 ?
I would recommend you join the IRC channel so we can help you in
realtime and get a clear understanding of what you're trying to
accomplish.
Thanks,
Brian
>
> Thanks
> Paul.
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Brian West
sip:brian at freeswitch.org
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