[Freeswitch-users] Need help with a gateway problem

Pete Kay petedao at gmail.com
Thu Apr 17 08:07:40 PDT 2008


Hi,

I tried "sofia status" and it does not show my gateway,

freeswitch at ser> sofia status
API CALL [sofia(status)] output:
                     Name          Type
Data      State
=================================================================================================
            192.168.1.104         alias
default      ALIASED
                 outbound       profile   sip:mod_sofia at 58.251.75.228:5080
RUNNING (0)
                  default       profile   sip:mod_sofia at 192.168.1.104:5060
RUNNING (0)
                      nat       profile   sip:mod_sofia at 58.251.75.228:5070
RUNNING (0)
=================================================================================================
3 profiles 1 alias

I suspected maybe something is wrong with my config but I just can't figure
it out.

Could you please help to take a look at my pastebin?
http://pastebin.freeswitch.org/4248

Please help me so I get get started with FS.

Thanks,
Pete

On Thu, Apr 17, 2008 at 10:48 PM, UV <uv at talknet.com.au> wrote:

>  Try this instead:
>
> <action application="bridge" data="sofia/gateway/callwithus/$1@
> sip.callwithus.com"/>
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Pete Kay
> *Sent:* Thursday, April 17, 2008 11:02 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Need help with a gateway problem
>
>
>
> Hi,
>
> I am still not yet able to get one sip phone to call the other due to the
> problem I posted.  So, I used the working sip client to try to call an
> outside number
> by setting up a gateway.
>
> I tried to set up fastswitch to route call to my voip provider, I am
> getting channel error but can't know why.  Can anyone please give some help?
>
> this is in my sofia.conf.xml
>
>
>
> <gateways>
>
>          <gateway name="callwithus">
>
>            <!--/// account username *required* ///-->
>
>            <param name="username" value="226"/>
>
>            <!--/// auth realm: *optional* same as gateway name, if blank
> ///-->
>
>            <param name="realm" value="callwithus"/>
>
>            <!--/// account password *required* ///-->
>
>            <param name="password" value="3336"/>
>
>            <!--/// extension for inbound calls: *optional* same as
> username, if$           <param name="extension" value="2"/>
>
>            <!--/// proxy host: *optional* same as realm, if blank ///-->
>
>            <param name="proxy" value="sip.callwithus.com"/>
>
>            <!--/// expire in seconds: *optional* 3600, if blank ///-->
>
>            <param name="expire-seconds" value="600"/>
>
>
>
>          </gateway>
>
>        </gateways>
>
>
>
>
>
>
>
>
>
> this is in my default dialplan
>
>
>
>  <context name="home">
>
>    <extension name="PrintVars" continue="true">
>
>     <condition field="destination_number" expression="^[0-9]">
>
>      <action application="info"/>
>
>     </condition>
>
>    </extension>
>
>
>
>    <extension name="PBX Extension">
>
>     <condition field="destination_number" expression="^(1[0-9]{2})$">
>
>      <!-- This will dial a registered phone at the ip or domain name you
> set in$     <!-- (The % indicates it is an internal extension)
>             $     <action application="bridge" data="sofia/sip/$1%${
> server-domain-name}"/>
>
>     </condition>
>
>    </extension>
>
>   <extension name="Local Dial">
>
>     <condition field="destination_number" expression="^([0-9]{13})$">
>
>      <action application="set" data="effective_caller_id_name=John
> Freeswitch"/>     <action application="set"
> data="effective_caller_id_number=1234567"/>
>
>         <action application="log" data="before bridging [${1}]\n"/>
>
>      <action application="bridge" data="sofia/gateway/callwithus/$1 at callwithus"$
>   </condition>
>
>    </extension>
>
>   </context>
>
>
>
>
>
> Get error when I dial a 13 digits number:
>
>
>
> 2008-04-18 04:26:03 [ERR] mod_sofia.c:1692 sofia_outgoing_channel()Invalid Gateway
>
> 2008-04-18 04:26:03 [NOTICE] mod_sofia.c:1857 sofia_outgoing_channel()Close Channel N/A
> [CS_NEW]
>
> 2008-04-18 04:26:03 [ERR] switch_ivr_originate.c:813 switch_ivr_originate
> () Cannot create outgoing channel of type [sofia] cause: [
> INVALID_NUMBER_FORMAT]
>
> 2008-04-18 04:26:03 [INFO] mod_dptools.c:1536 audio_bridge_function()Originate Failed.  Cause: INVALID_NUMBER_FORM AT
>
>
> Any suggestion will be greatly appreciated.
>
> Thanks
> Pete
>
>
>
> No virus found in this outgoing message.
> Checked by AVG.
> Version: 7.5.519 / Virus Database: 269.23.0/1382 - Release Date: 16-Apr-08
> 5:34 PM
>
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