effective_caller_id_name in user directory not being passed to SIP endpoint
Hi all, I'm starting out with FS 1.4.7 on CentOS 6 (installed via the FS repo at http://files.freeswitch.org) and have a simple working config with two sofia profiles, internal and external. I can register SIP clients, call between internal extensions and call external numbers through a SIP gateway. Each user in directory/default/*.xml looks like this: <include> <user id="106"> <params> <param name="password" value="xxx"/> </params> <variables> <variable name="accountcode" value="106"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Gavin Hamill"/> <variable name="outbound_caller_id_name" value="Gavin Hamill"/> </variables> </user> </include> However, when I call from 106 -> 105, I only see '105' appear in the destination client when I would expect to see 'Gavin Hamill' If I force the effective_caller_id_name to be a fixed string: <context name="default"> <extension name="bashton_office_staff"> <condition field="destination_number" expression="^(10[0-9])quot;> <action application="set" data="effective_caller_id_name=QWERTYUIOP" /> <action application="bridge" data="user/$1"/> </condition> ... </context> Then sure enough calls always appear as coming from 'QWERTYUIOP' At https://confluence.freeswitch.org/display/FREESWITCH/XML+User+Directory there is a section 'Variables' which states: Any variables defined in the domain or user will be defined as channel variables when there is a call to user or when there is an inbound calls from that user. That's exactly the behaviour I was expecting - any pointers where I'm going wrong would be warmly received. Cheers, Gavin
Sounds like you may have turned off auth calls and its never actually setting anything and just letting the calls thru. On Tue, Sep 9, 2014 at 8:03 AM, Gavin Hamill <gavin@bashton.com> wrote:
Hi all,
I'm starting out with FS 1.4.7 on CentOS 6 (installed via the FS repo at http://files.freeswitch.org) and have a simple working config with two sofia profiles, internal and external. I can register SIP clients, call between internal extensions and call external numbers through a SIP gateway.
Each user in directory/default/*.xml looks like this:
<include> <user id="106"> <params> <param name="password" value="xxx"/> </params> <variables> <variable name="accountcode" value="106"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Gavin Hamill"/> <variable name="outbound_caller_id_name" value="Gavin Hamill"/> </variables> </user> </include>
However, when I call from 106 -> 105, I only see '105' appear in the destination client when I would expect to see 'Gavin Hamill' If I force the effective_caller_id_name to be a fixed string:
<context name="default"> <extension name="bashton_office_staff"> <condition field="destination_number" expression="^(10[0-9])quot;> <action application="set" data="effective_caller_id_name=QWERTYUIOP" /> <action application="bridge" data="user/$1"/> </condition> ... </context>
Then sure enough calls always appear as coming from 'QWERTYUIOP' At https://confluence.freeswitch.org/display/FREESWITCH/XML+User+Directory there is a section 'Variables' which states:
Any variables defined in the domain or user will be defined as channel variables when there is a call to user or when there is an inbound calls from that user.
That's exactly the behaviour I was expecting - any pointers where I'm going wrong would be warmly received.
Cheers, Gavin
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Brian, Yes you're right - I started from almost zero config so auth-calls was not set in the sofia profile. I solved the issue yesterday by using set_user based on the Caller ID (i.e. trusting the incoming Caller ID for the 'default' context' - accessible by internal port 5060 calls only). It turns out that whilst I already had this in vars.xml: <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> I didn't provide the matching line in sip_profiles/internal.xml: <param name="auth-calls" value="$${internal_auth_calls}"/> I have now added that line to the sip profile, removed the 'set_user' from the dialplan, and authentication is working just as it should. Many thanks for taking the time to point me in the right direction - it's really appreciated! Cheers, Gavin. On 9 September 2014 16:32, Brian West <brian@freeswitch.org> wrote:
Sounds like you may have turned off auth calls and its never actually setting anything and just letting the calls thru.
On Tue, Sep 9, 2014 at 8:03 AM, Gavin Hamill <gavin@bashton.com> wrote:
Hi all,
I'm starting out with FS 1.4.7 on CentOS 6 (installed via the FS repo at http://files.freeswitch.org) and have a simple working config with two sofia profiles, internal and external. I can register SIP clients, call between internal extensions and call external numbers through a SIP gateway.
Each user in directory/default/*.xml looks like this:
<include> <user id="106"> <params> <param name="password" value="xxx"/> </params> <variables> <variable name="accountcode" value="106"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Gavin Hamill"/> <variable name="outbound_caller_id_name" value="Gavin Hamill"/> </variables> </user> </include>
However, when I call from 106 -> 105, I only see '105' appear in the destination client when I would expect to see 'Gavin Hamill' If I force the effective_caller_id_name to be a fixed string:
<context name="default"> <extension name="bashton_office_staff"> <condition field="destination_number" expression="^(10[0-9])quot;> <action application="set" data="effective_caller_id_name=QWERTYUIOP" /> <action application="bridge" data="user/$1"/> </condition> ... </context>
Then sure enough calls always appear as coming from 'QWERTYUIOP' At https://confluence.freeswitch.org/display/FREESWITCH/XML+User+Directory there is a section 'Variables' which states:
Any variables defined in the domain or user will be defined as channel variables when there is a call to user or when there is an inbound calls from that user.
That's exactly the behaviour I was expecting - any pointers where I'm going wrong would be warmly received.
Cheers, Gavin
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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Brian West
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Gavin Hamill