[Freeswitch-users] outbound call, new comer

[Will Smith]

Forgive me,
I don't know how to turn on the SIP debug mode. This is what it say from FS command line:

2009-01-13 16:26:46 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing 1000->18187188288in context default
2009-01-13 16:26:47 [ERR] mod_sofia.c:2341 sofia_outgoing_channel() Invalid Gateway
2009-01-13 16:26:47 [NOTICE] mod_sofia.c:2540 sofia_outgoing_channel() Close Channel N/A [CS_NEW]
2009-01-13 16:26:47 [ERR] switch_ivr_originate.c:1110 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2009-01-13 16:26:47 [INFO] mod_dptools.c:1891 audio_bridge_function() Originate Failed.  Cause: INVALID_NUMBER_FORMAT
2009-01-13 16:26:47 [NOTICE] mod_dptools.c:1918 audio_bridge_function() Hangup sofia/internal/1000 at 192.168.2.104 [CS_EXECUTE] [INVALID_NUMBER_FORMAT]
2009-01-13 16:26:47 [NOTICE] switch_core_session.c:956 switch_core_session_thread() Session 163 (sofia/internal/1000 at 192.168.2.104) Ended
2009-01-13 16:26:47 [NOTICE] switch_core_session.c:958 switch_core_session_thread() Close Channel sofia/internal/1000 at 192.168.2.104 [CS_HANGUP]


--- On Mon, 1/12/09, Kristian Kielhofner <kristian.kielhofner at gmail.com> wrote:
From: Kristian Kielhofner <kristian.kielhofner at gmail.com>
Subject: Re: [Freeswitch-users] outbound call, new comer
To: freeswitch-users at lists.freeswitch.org
Date: Monday, January 12, 2009, 1:07 PM

In West Philadelphia born and raised...

Voicepulse seems to be picky about number format.  Trying doing full
E.164 (+1).  Also, make sure your realm is correct.  What does a SIP
debug look like?

On 1/12/09, Will Smith <willbelair at yahoo.com> wrote:
>
>
> Hi,
> I am first time FS user, so it is a bit confused with all the setup. For
inbound calls, I tried to add a voicepulse.xml in the sip_profiles/external with
the following codes:
>
> <include>
>   <gateway name="voicepulse">
>     <param name="username" value="3334445555"/>
>     <param name="realm"
value="my_sip_provider.com"/>
>     <param name="password" value="3334445555"/>
>     <param name="proxy"
value="my_sip_provider.com"/>
>     <param name="expire-seconds" value="600"/>
>     <param name="register" value="true"/>
>   </gateway>
>   <gateway name="voicepulse-backup">
>     <param name="username" value="3334445555"/>
>     <param name="realm"
value="my_sip_provider.com"/>
>     <param name="password" value="3334445555"/>
>     <param name="extension" value="3334445555"/>
>     <param name="proxy" 
value="my_sip_provider.com"/>
>     <param name="expire-seconds" value="600"/>
>     <param name="register" value="true"/>
>   </gateway>
> </include>
>
> ------------
>  and in the conf/dialplan/default.xml file I added:
>
> <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444)
here -->
>    <extension name="Long Distance - voicepulse">
>     <condition field="destination_number"
expression="^(1{0,1}\d{10})$">
>       <action application="set"
data="effective_caller_id_number=12223334444"/>
>       <!-- If your provider does not provide ringback (180 or 183) you
may simulate
>         ringback by uncommenting the following line. -->
>       <!-- action application="ringback" /-->
>       <action application="bridge"
data="sofia/gateway/voicepulse/$1"/>
>      </condition>
>    </extension>
>
>
> ------------------
>
> For inbound, I added
>
> <extension name="Voicepulse">   <!-- your provider or
any name you'd like to call it -->
>     <condition field="destination_number"
expression="3334445555">  <!-- your DID for this gateway-->
>        <action application="transfer" data="1001 XML
default"/>
>     </condition>
>    </extension>
>
> ----------
>
> And, if I dial 3334445555 from a softphone registered with
my_sip_provider, I got the message to the voice mail of 1001 - the 1001
extension does not ring.
> And if from 1001, I dial some real number like 18188892345, I got the
error: Invalid Gateway ... Cannot create outgoing channel of type [fosia] cause:
[Invalid_number_format] ...
>
>
> Would someone please give me some help to set this up. I am a bit confused
with these.
>
> Thank you
>
> _______________________________________________
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>  Freeswitch-users at lists.freeswitch.org
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>
>



-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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