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

Michael Collins msc at freeswitch.org
Wed May 30 00:29:32 MSD 2012 

Sean,

Thanks for taking the time to follow up. If you get it working we'd love to
have a snapshot of your configuration to put up on the wiki.

Thanks,
MC

On Tue, May 29, 2012 at 1:19 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:

> Thanks Anthony.
>
> First I should have pointed out that the 2 test phones are on the local Lan
> with the switch, NO NAT.  The final task is to move them out with NAT.
>
> Second, it is working!!
>
> After I dumped everything for you I noticed one of the phones was
> connecting
> on the wrong port (sofia external, not lan).  I changed it last week but it
> did not seem to work.  Now having waited (and registrations having renewed)
> it is working!!
>
> Thank you for your help.
>
> Now, out to the site and their crappy router and to see if this works on
> through NAT.
>
> Sean
> -----Original Message-----
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
> Sent: Tuesday, May 29, 2012 3:18 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain
>
> 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}${sofia_contact(220 at mydomain
> >> n
> >> ame.com)}"/>
> >>
> >>         <param name="mwi-account" value="220 at voicemail_BFIS"/>
> >>
> >>       </params>
> >>
> >>     </user>
> >>
> >>
> >>
> >> And the dial plan for ext 220:
> >>
> >>   <extension name="220" >
> >>
> >>     <condition field="destination_number" expression="^220$">
> >>
> >>       <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_inf
> >> o
> >> 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:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> Join Us At ClueCon - Aug 7-9, 2012
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> >> e
> >> rs
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > Join Us At ClueCon - Aug 7-9, 2012
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
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