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

Anthony Minessale anthony.minessale at gmail.com
Tue May 29 23:17:48 MSD 2012 

enter "sofia_contact 220 at mydomainname.com" at your cli and see if you
get a large dial string.
if you are only getting calls to one phone make sure they all work,
its possible you have many phones behind the same nat and only one of
them can poke through at a time.



On Tue, May 29, 2012 at 12:11 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> BUMP!
>
> Anyone have any ideas for me?
>
> Any other information I can provide?
>
> Thanks.
> Sean
>
> -----Original Message-----
> From: Sean Devoy [mailto:sdevoy at bizfocused.com]
> Sent: Friday, May 25, 2012 6:06 PM
> To: 'FreeSWITCH Users Help'
> Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain
>
> Here is the result of select * from sip_subscriptions;"
> sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user"
> <sip:220 at 10.10.40.30:5060>|38e107ab-6cde6635 at 10.10.40.30|"220"
> <sip:220 at fs_lan.bizfocused.com>;tag=fd508933c5f5924d|SIP/2.0/UDP
> 10.10.40.30:5060;branch=z9hG4bK-f5598bfb|1337980638|Cisco/SPA504G-7.4.8a||ex
> ternal|FreeSwitch1.sumdomain.com|5060|10.10.40.30|-1||"220"
> <sip:220 at fs_lan.bizfocused.com>;tag=VrGrXQaOH22R
> sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user"
> <sip:220 at 10.10.40.20:5064>|c08f0c6a-c46e90d2 at 10.10.40.20|"Sean"
> <sip:220 at fs_lan.bizfocused.com>;tag=f22a978ae8838032|SIP/2.0/UDP
> 10.10.40.20:5064;branch=z9hG4bK-905b6faa|1337980643|Cisco/SPA504G-7.4.9c||ex
> ternal|FreeSwitch1.sumdomain.com|5064|10.10.40.20|-1||"Sean"
> <sip:220 at fs_lan.bizfocused.com>;tag=y5VtigPlIghD
>
> But it was in:
> sofia_reg_external.db not internal.
>
> I have sorted out all the sip trace data into 2 txt files for the 2 phones
> involved.  They are zipped up at:
> http://www.bizfocused.com/sip_trace.zip
>
> Thank you again for your help.  I am way over my head now.
>
>
> -----Original Message-----
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> Sent: Friday, May 25, 2012 2:35 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain
>
> 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}${sofia_contact(220 at mydomainn
>> ame.com)}"/>
>>
>>         <param name="mwi-account" value="220 at voicemail_BFIS"/>
>>
>>       </params>
>>
>>     </user>
>>
>>
>>
>> And the dial plan for ext 220:
>>
>>   <extension name="220" >
>>
>>     <condition field="destination_number" expression="^220$">
>>
>>       <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
>>
>>
>> ______________________________________________________________________
>> ___ Professional FreeSWITCH Consulting Services:
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>
>
>
> --
> Anthony Minessale II
>
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>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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> http://www.cluecon.com
>
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>
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-- 
Anthony Minessale II

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