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

Anthony Minessale anthony.minessale at gmail.com
Wed May 30 00:31:22 MSD 2012


They should work either way.

On Tue, May 29, 2012 at 3:28 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> One follow up question about SCA and Gateways ...
>
> I have one Gateway setup w/out NAT for phones on the local LAN and a second
> for NAT phones (on a different port). Is that necessary or can I support NAT
> and NO NAT phones on the same Gateway/port?  If 2 ports/gateways is the best
> approach will SCA work across them?  I don’t see how as one would be
> 220 at local_ip_name.com and other would be 220 at external_ip_name.com.
>
> Thanks again,
> 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
>
> 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-users
> 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



Join us at ClueCon 2011 Aug 9-11, 2011
More information about the FreeSWITCH-users mailing list