[Freeswitch-users] SCA not working inbound - Multi Domain

[Anthony Minessale]

What are the phones putting in the subscribe ?

sofia global siptrace on
sofia global debug presence|sla

then watch for SUBSCRIBE

also when you are not using odbc you can get the sql with this app

sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db

also try "select * from sip_subscriptions"

its all about using the right host name across the board, IP's count
as hostnames, they do not magically resolve any dns with SIP




On Fri, May 25, 2012 at 1:26 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> Hi all,
>
>
>
> I have a muti-tennnant configuration that is working nicely except for
> Shared Call Appearance.  The desktop devices are CISCO 504Gs and they are
> configured as described in the FS Wiki as well as Cisco Documentation.
>
>
>
> The SCA works perfectly for outbound calls – if either phone pickups like
> 220, the other phones indicator light flashes red.  However, inbound calls
> will go to only one of the phones (which one has changed a few times) and
> the other phones line still just stays green and does not ring.
>
>
>
> Here is the sip interfaces config:
>
> <profile name="internal">
>
>     <settings>
>
>       <param name="enable-timer" value="false"/>
>
>       <param name="user-agent-string" value="Configured by Sean!"/>
>
>       <param name="rtp-timer-name" value="soft"/>
>
>       <param name="codec-prefs" value="$${global_codec_prefs}"/>
>
>       <param name="manage-shared-appearance" value="true"/>
>
>       <param name="multiple-registrations" value="true"/>
>
>       <param name="manage-presence" value="true"/>
>
>       <param name="inbound-codec-negotiation" value="generous"/>
>
>       <param name="inbound-reg-force-matching-username" value="true"/>
>
>       <param name="nonce-ttl" value="86400"/>
>
>       <param name="rfc2833-pt" value="101"/>
>
>       <param name="manage-presence" value="true"/>
>
>       <param name="auth-calls" value="true"/>
>
>       <param name="sip-ip" value="10.10.40.185"/>
>
>       <param name="rtp-ip" value="10.10.40.185"/>
>
>       <param name="sip-port" value="5064"/>
>
>       <param name="nat-options-ping" value="false"/>
>
>       <param name="all-reg-options-ping" value="true"/>
>
>       <param name="context" value="from-BFIS"/>
>
>     </settings>
>
>   </profile>
>
>
>
> The directory entry which both phones connect using:
>
>     <user id="220">
>
>       <variables>
>
>         <variable name="outbound_caller_id_name" value="BFIS Sean"/>
>
>         <variable name="outbound_caller_id_number" value="410420BLEEP"/>
>
>         <variable name="internal_caller_id_name" value="Sean BLEEP"/>
>
>         <variable name="internal_caller_id_number" value="220"/>
>
>         <variable name="user_context" value="from-internal-BFIS"/>
>
>         <variable name="user_originated" value="true"/>
>
>         <variable name="toll_allow" value="domestic"/>
>
>         <variable name="accountcode" value="220"/>
>
>         <variable name="mwi-account" value="220 at voicemail_BFIS"/>
>
>       </variables>
>
>       <params>
>
>         <param name="manage-shared-appearance" value-="true" />"
>
>         <param name="password"  value="BLEEPBLEEP"/>
>
>         <param name="dial-string"
> value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomainname.com)}"/>
>
>         <param name="mwi-account" value="220 at voicemail_BFIS"/>
>
>       </params>
>
>     </user>
>
>
>
> And the dial plan for ext 220:
>
>   <extension name="220" >
>
>     <condition field="destination_number" expression="^220$">
>
>       <action application="set"
> data="effective_caller_id_number=${internal_caller_id_number}"/>
>
>       <action application="set"
> data="effective_caller_id_name=${internal_caller_id_name}"/>
>
>       <action application="set" data="ringback=${us-ring}"/>
>
>       <action application="set" data="call_timeout=20"/>
>
>       <action application="set" data="hangup_after_bridge=true"/>
>
>       <action application="bridge"
> data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com"  />
>
>       <action application="answer"/>
>
>       <action application="voicemail" data="default voicemail_BFIS 220"/>
>
>       <action application="hangup"/>
>
>     </condition>
>
>   </extension>
>
>
>
>
>
>
>
> I did see this in the wiki
> (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance):
>
> If SLA works for outgoing calls and SLA does not work for inbound calls to
> the SLA phones, you may have some presence problem related to mixed IP and
> domain names. When using ODBC you may issue the following SQL statement
>
> select
> sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_info from
> sip_dialogs;
>
> But I don’t have ODBC on this server, so I am a little lost.
>
>
>
> I have the phones login to domain names, not addresses.  I never refer to IP
> addresses in my xml (except gateways addresses).  I am not trying SLA across
> domain, only within the same domain.
>
>
>
> I hope someone can spot something.  Thanks for your help.
>
>
>
> Sean
>
>
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> 
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-- 
Anthony Minessale II

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