I've got the following SIP gateway which registers successfully at the ITSP: <gateway name="sipcall_41325133196"> <param name="username" value="41325133196"/> <param name="password" value="XXXXXXXXXX"/> <param name="extension" value="0325133196"/> <param name="expire-seconds" value="125"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="proxy" value="free2.voipgateway.org"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="false"/> <param name="ping" value="27"/> <variables> <variable name="sbc_callee_ccode" data="41" direction="inbound"/> <variable name="sbc_caller_ccode" data="41" direction="inbound"/> </variables> </gateway> Then the call from PSTN goes into the proper context, and the first condition in the context is: <extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> <action application="log" data="INFO Country code: ${sbc_callee_ccode}"/> </condition> </extension> Neither the info, nor the log show my two variables on inbound calls. Am I doing something wrong or is it a bug? I'm using git commit 94664868a8 in v1.2.stable branch thanks, stanislav
Several silly questions to help start debugging: 1) Did you kill the gateway and restart it since adding in this variable? reloadxml doesn't update active gateways. 2) Are you sure the call is actually coming in through the gateway and not directly to the sip address? 3) Is the debug extension actually executing? Are you sure ${call_debug}==true -- that lines runs in your logs? -Avi On Thu, Nov 1, 2012 at 11:00 PM, Stanislav Sinyagin <ssinyagin@yahoo.com>wrote:
I've got the following SIP gateway which registers successfully at the ITSP:
<gateway name="sipcall_41325133196"> <param name="username" value="41325133196"/> <param name="password" value="XXXXXXXXXX"/> <param name="extension" value="0325133196"/> <param name="expire-seconds" value="125"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="proxy" value="free2.voipgateway.org"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="false"/> <param name="ping" value="27"/> <variables> <variable name="sbc_callee_ccode" data="41" direction="inbound"/> <variable name="sbc_caller_ccode" data="41" direction="inbound"/> </variables> </gateway>
Then the call from PSTN goes into the proper context, and the first condition in the context is:
<extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> <action application="log" data="INFO Country code: ${sbc_callee_ccode}"/> </condition> </extension>
Neither the info, nor the log show my two variables on inbound calls.
Am I doing something wrong or is it a bug?
I'm using git commit 94664868a8 in v1.2.stable branch
thanks, stanislav
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[the mailing list is very slow lately, but I've seen some replies in the list archive]. freeswitch daemon was freshly started after XML editing. Also I'm sure the call comes into this gateway, as I get variable_sip_gateway: [sipcall_41325133196] in the info output. The full log for the call is here: https://gist.github.com/3997100 the call does not end anywhere, as there are no matching conditions yet that would end in a transfer or a bridge, so the caller gets the busy signal. But anyway I believe info should display the inbound variables. ----- Original Message -----
From: Stanislav Sinyagin <ssinyagin@yahoo.com> To: Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, November 1, 2012 10:00 PM Subject: Sofia gateway variables
I've got the following SIP gateway which registers successfully at the ITSP:
<gateway name="sipcall_41325133196"> <param name="username" value="41325133196"/> <param name="password" value="XXXXXXXXXX"/> <param name="extension" value="0325133196"/> <param name="expire-seconds" value="125"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="proxy" value="free2.voipgateway.org"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="false"/> <param name="ping" value="27"/> <variables> <variable name="sbc_callee_ccode" data="41" direction="inbound"/> <variable name="sbc_caller_ccode" data="41" direction="inbound"/> </variables> </gateway>
Then the call from PSTN goes into the proper context, and the first condition in the context is:
<extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> <action application="log" data="INFO Country code: ${sbc_callee_ccode}"/> </condition> </extension>
Neither the info, nor the log show my two variables on inbound calls.
Am I doing something wrong or is it a bug?
I'm using git commit 94664868a8 in v1.2.stable branch
thanks, stanislav
actually here's the whole FreeSWITCH config, except for the SIP gateway with the real password: https://github.com/xlab1/sipfe_freeswitch_common It's pretty short currently, and I re-define some global variables with X-PRE-PROCESS statements, so there is a potentially problem with that, such as corrupted memory. Particularly, I define the IPv4 addresses in vars.xml, and set them again with the proper values in site_vars.xml I don't know if the pre-processor is designed to handle this, but it works so far... ----- Original Message -----
From: Stanislav Sinyagin <ssinyagin@yahoo.com> To: Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, November 1, 2012 11:31 PM Subject: Re: Sofia gateway variables
[the mailing list is very slow lately, but I've seen some replies in the list archive].
freeswitch daemon was freshly started after XML editing. Also I'm sure the call comes into this gateway, as I get variable_sip_gateway: [sipcall_41325133196] in the info output.
The full log for the call is here: https://gist.github.com/3997100
the call does not end anywhere, as there are no matching conditions yet that would end in a transfer or a bridge, so the caller gets the busy signal. But anyway I believe info should display the inbound variables.
----- Original Message -----
From: Stanislav Sinyagin <ssinyagin@yahoo.com> To: Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, November 1, 2012 10:00 PM Subject: Sofia gateway variables
I've got the following SIP gateway which registers successfully at the ITSP:
<gateway name="sipcall_41325133196"> <param name="username" value="41325133196"/> <param name="password" value="XXXXXXXXXX"/> <param name="extension" value="0325133196"/> <param name="expire-seconds" value="125"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="proxy" value="free2.voipgateway.org"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="false"/> <param name="ping" value="27"/> <variables> <variable name="sbc_callee_ccode" data="41" direction="inbound"/> <variable name="sbc_caller_ccode" data="41" direction="inbound"/> </variables> </gateway>
Then the call from PSTN goes into the proper context, and the first condition in the context is:
<extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> <action application="log" data="INFO Country code:
${sbc_callee_ccode}"/> </condition> </extension>
Neither the info, nor the log show my two variables on inbound calls.
Am I doing something wrong or is it a bug?
I'm using git commit 94664868a8 in v1.2.stable branch
thanks, stanislav
Where are you setting call_debug=true? Are you doing it before a transfer? Maybe you need to use inline? http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions On Thu, Nov 1, 2012 at 5:00 PM, Stanislav Sinyagin <ssinyagin@yahoo.com> wrote:
I've got the following SIP gateway which registers successfully at the ITSP:
<gateway name="sipcall_41325133196"> <param name="username" value="41325133196"/> <param name="password" value="XXXXXXXXXX"/> <param name="extension" value="0325133196"/> <param name="expire-seconds" value="125"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="proxy" value="free2.voipgateway.org"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="false"/> <param name="ping" value="27"/> <variables> <variable name="sbc_callee_ccode" data="41" direction="inbound"/> <variable name="sbc_caller_ccode" data="41" direction="inbound"/> </variables> </gateway>
Then the call from PSTN goes into the proper context, and the first condition in the context is:
<extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> <action application="log" data="INFO Country code: ${sbc_callee_ccode}"/> </condition> </extension>
Neither the info, nor the log show my two variables on inbound calls.
Am I doing something wrong or is it a bug?
I'm using git commit 94664868a8 in v1.2.stable branch
thanks, stanislav
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
-- Kristian Kielhofner
it is set in the usual place, in vars.xml as a global variable. Besides, I do get the full Info output, but those gateway variables are missing: https://gist.github.com/3997100 ----- Original Message -----
From: Kristian Kielhofner <kris@kriskinc.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Friday, November 2, 2012 5:06 PM Subject: Re: [Freeswitch-users] Sofia gateway variables
Where are you setting call_debug=true? Are you doing it before a transfer? Maybe you need to use inline?
http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
On Thu, Nov 1, 2012 at 5:00 PM, Stanislav Sinyagin <ssinyagin@yahoo.com> wrote:
I've got the following SIP gateway which registers successfully at the ITSP:
<gateway name="sipcall_41325133196"> <param name="username" value="41325133196"/> <param name="password" value="XXXXXXXXXX"/> <param name="extension" value="0325133196"/> <param name="expire-seconds" value="125"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="proxy" value="free2.voipgateway.org"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="false"/> <param name="ping" value="27"/> <variables> <variable name="sbc_callee_ccode" data="41" direction="inbound"/> <variable name="sbc_caller_ccode" data="41" direction="inbound"/> </variables> </gateway>
Then the call from PSTN goes into the proper context, and the first condition in the context is:
<extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> <action application="log" data="INFO Country code: ${sbc_callee_ccode}"/> </condition> </extension>
Neither the info, nor the log show my two variables on inbound calls.
Am I doing something wrong or is it a bug?
I'm using git commit 94664868a8 in v1.2.stable branch
thanks, stanislav
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
Problem is resolved: http://jira.freeswitch.org/browse/FS-4791 Wiki documented a wrong syntax (data="xx" instead of value="xx"), and I updated the wiki pages where I could find it. ----- Original Message -----
From: Stanislav Sinyagin <ssinyagin@yahoo.com> To: Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, November 1, 2012 10:00 PM Subject: [Freeswitch-users] Sofia gateway variables
I've got the following SIP gateway which registers successfully at the ITSP:
<gateway name="sipcall_41325133196"> <param name="username" value="41325133196"/> <param name="password" value="XXXXXXXXXX"/> <param name="extension" value="0325133196"/> <param name="expire-seconds" value="125"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="proxy" value="free2.voipgateway.org"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="false"/> <param name="ping" value="27"/> <variables> <variable name="sbc_callee_ccode" data="41" direction="inbound"/> <variable name="sbc_caller_ccode" data="41" direction="inbound"/> </variables> </gateway>
Then the call from PSTN goes into the proper context, and the first condition in the context is:
<extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> <action application="log" data="INFO Country code: ${sbc_callee_ccode}"/> </condition> </extension>
Neither the info, nor the log show my two variables on inbound calls.
Am I doing something wrong or is it a bug?
I'm using git commit 94664868a8 in v1.2.stable branch
thanks, stanislav
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
Eek, looks like I doc'd that. I can't find where I had copied it from, I must have simply copied it from the dialplan, which uses "data" instead of "value". Sorry for the trouble and thanks for tracking it down! -Avi On Sat, Nov 3, 2012 at 11:32 PM, Stanislav Sinyagin <ssinyagin@yahoo.com>wrote:
Problem is resolved: http://jira.freeswitch.org/browse/FS-4791
Wiki documented a wrong syntax (data="xx" instead of value="xx"), and I updated the wiki pages where I could find it.
----- Original Message -----
From: Stanislav Sinyagin <ssinyagin@yahoo.com> To: Free SWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Sent: Thursday, November 1, 2012 10:00 PM Subject: [Freeswitch-users] Sofia gateway variables
I've got the following SIP gateway which registers successfully at the ITSP:
<gateway name="sipcall_41325133196"> <param name="username" value="41325133196"/> <param name="password" value="XXXXXXXXXX"/> <param name="extension" value="0325133196"/> <param name="expire-seconds" value="125"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="proxy" value="free2.voipgateway.org"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="false"/> <param name="ping" value="27"/> <variables> <variable name="sbc_callee_ccode" data="41" direction="inbound"/> <variable name="sbc_caller_ccode" data="41" direction="inbound"/> </variables> </gateway>
Then the call from PSTN goes into the proper context, and the first condition in the context is:
<extension name="call_debug" continue="true"> <condition field="${call_debug}" expression="^true$" break="never"> <action application="info"/> <action application="log" data="INFO Country code: ${sbc_callee_ccode}"/> </condition> </extension>
Neither the info, nor the log show my two variables on inbound calls.
Am I doing something wrong or is it a bug?
I'm using git commit 94664868a8 in v1.2.stable branch
thanks, stanislav
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 (4)
-
Avi Marcus
-
Kristian Kielhofner
-
Paul Cupis
-
Stanislav Sinyagin