[Freeswitch-users] inbound calls
Ilan Perez
iperez at diagnosticdev.com
Mon Aug 4 22:29:04 PDT 2008
I have had inbound calls working for about a day.I am finding now that after
1min 41 seconds the call automatically hangs up.
Any ideas why?
Ilan Perez
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Roberto
Hernandez
Sent: 05 August 2008 12:54
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] inbound calls
Three things. debug mode is a beautiful thing.
First, my mistake:
I had set my trunk (peer) user & friendly name to 'main'.
I modified these on my SIP provider portal, called my DID, and I was able to
pass the dial plan step.
For this example say my DID is 3213214321.
[DEBUG] mod_dialplan_xml.c:107 parse_exten() test conditions
destination_number(main) =~ /^(3213214321)$/
where DID is my SIP provider number.
Second, I added this to: conf/dialplan/public.xml
<extension name="test_did">
<condition field="destination_number" expression="^(3213214321)$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
Modified to transfer to 1000 instead of $1 (my SIP endpoint
conf/directory/default/1000.xml), else I would have had to change my user
id.
All depend how you want to organize endpoints.
<extension name="test_did">
<condition field="destination_number" expression="^(3213214321)$">
<action application="transfer" data="1000 XML default"/>
</condition>
</extension>
Third, I added this to: conf/dialplan/default.xml
<!-- Inbound calls handled first - You will want to configure one or
-->
<!-- more of these depending upon the ways in which you can receive calls
-->
<!-- If you have a number of similar DID's and they get the same call
treatment -->
<!-- you may want to use Regex pattern matching instead of a hard coded
number -->
<extension name="Inbound-3213214321">
<!-- EDIT: change the DID to your inbound DID (DN) number -->
<!-- Note - you can use regx pattern matching if needed -->
<condition field="destination_number" expression="^3213214321$">
<!-- If you are going to ring multiple extensions you should send back
-->
<!-- a 180 ringing message to the provider.
-->
<action application="ringback" />
<!-- Set the maximum amount of time you want to ring the extensions
(seconds) -->
<action application="set" data="call_timeout=20"/>
<!-- Sample: Bridge the call to 100, 101, 102, 103, 104 and 105
extensions. If you -->
<!-- do not have 5 extensions configured you can remove the extra ones
or just leave -->
<!-- them in... it will not cause a problem... just extra log messages.
-->
<!-- Note: use "," between extensions to dial at the same time and "|"
to dial sequentially. -->
<!-- Note: No spaces can exist in the data string
data="sofia/sip/100%${...},sofia/sip/101%${...} -->
<!-- EDIT: You will also need to change the IP (or domain name) in the
URI to that of your server. -->
<action application="bridge" data="sofia/sip/1000%$${domain}"/>
<!-- No one answered so launch the answering machine application -->
<action application="javascript"
data="/usr/local/freeswitch/scripts/pizza.js"/>
<!-- Sample single extension bridge -->
<!--action application="bridge" data="sofia/sip/1000%$${domain}"/-->
</condition>
</extension>
Thanks,
Roberto
_____
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ilan
Perez
Sent: Monday, August 04, 2008 6:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] inbound calls
What did you do?
Ilan Perez
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Roberto
Hernandez
Sent: 05 August 2008 08:53
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] inbound calls
Figured it out.
Thanks,
Roberto
_____
From: Roberto Hernandez [mailto:rh at bluevisor.com]
Sent: Monday, August 04, 2008 11:01 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: inbound calls
I've configured one X-Lite softphone (id=1000) and made my first few PSTN
outbound calls with no problem.
Inbound is a bit of a mystery. How do I configure PSTN inbound calls to
terminated on my softphone?
I found the following archived email, would someone break it down so new
users can understand? Thanks in advance.
In public.xml, you want something like this:
<extension name="test_did">
<condition field="destination_number" expression="^(4153084258)$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
and then in default.xml have something like this in the default context:
<extension name="Local_Extension">
<condition field="destination_number" expression="^(4153084258)$"
continue="on-true">
<action application="set" data="dialed_ext=$1"/>
</condition>
<condition field="destination_number"
expression="^${caller_id_number}$">
<action application="set"
data="voicemail_authorized=${sip_authorized}"/>
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="check default $${domain}
${dialed_ ext}"/>
<anti-action application="ring_ready"/>
<anti-action application="set" data="call_timeout=10"/>
<anti-action application="set" data="hangup_after_bridge=true"/>
<anti-action application="set" data="continue_on_fail=true"/>
<anti-action application="bridge"
data="USER/${dialed_ext}@$${domain}"/>
<anti-action application="answer"/>
<anti-action application="sleep" data="1000"/>
<anti-action application="voicemail" data="default $${domain}
${dialed_e xt}"/>
</condition>
</extension>
Thanks,
Roberto
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