Try:
(maybe
someone should add this to the wiki page http://wiki.freeswitch.org/wiki/Sofia
)
From:
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Pete Kay
Sent: Friday, April 18, 2008 1:08
AM
To:
Subject: Re: [Freeswitch-users]
Need help with a gateway problem
Hi,
I tried "
freeswitch@ser>
API CALL [
Name
Type
Data State
=================================================================================================
192.168.1.104
alias
default ALIASED
outbound profile sip:mod_sofia@58.251.75.228:5080
RUNNING (0)
default profile sip:mod_sofia@192.168.1.104:5060
RUNNING (0)
nat profile sip:mod_sofia@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@talknet.com.au> wrote:
Try this instead:
<action application="bridge"
data="sofia/gateway/callwithus/$1@sip.callwithus.com"/>
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Pete Kay
Sent: Thursday, April 17, 2008
11:02 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Need
help with a gateway problem
Hi,
I am still not yet able to get one sip phone to call the other due to the
problem I posted. So, I used the working sip client to try to call an
outside number
by setting up a gateway.
I tried to set up fastswitch to route call to my voip provider, I am getting
channel error but can't know why. Can anyone please give some help?
this is in my sofia.conf.xml
<gateways>
<gateway name="callwithus">
<!--/// account username *required* ///-->
<param name="username" value="226"/>
<!--/// auth realm: *optional* same as gateway name, if blank
///-->
<param name="realm" value="callwithus"/>
<!--/// account password *required* ///-->
<param name="password" value="3336"/>
<!--/// extension for inbound calls: *optional* same as username,
if$ <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/$1@callwithus"$
</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 [
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
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