Hi All, I have a vanilla debian 7 FreeSWITCH installation with a file called 201.xml in /etc/freeswitch/dialplan/default/ When I dial 201 (from extension 203) it won't ring the extension. I have included a SIP trace on: www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt Any ideas what I'm doing wrong? Thanks, as always, in advance Richard
HI Richard, it seems like your acl settings are blocking the incoming call. Is 203 registered with freeswitch? thanks Matt On 06/06/2014 07:47, Richard Mace wrote:
Hi All, I have a vanilla debian 7 FreeSWITCH installation with a file called 201.xml in /etc/freeswitch/dialplan/default/
When I dial 201 (from extension 203) it won't ring the extension. I have included a SIP trace on: www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt <http://www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt>
Any ideas what I'm doing wrong?
Thanks, as always, in advance
Richard
_________________________________________________________________________ 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
Hi Matt, I think so, what's the best command to check? Richard On 6 June 2014 09:42, Matt Broad <matt@inveroak.com> wrote:
HI Richard,
it seems like your acl settings are blocking the incoming call. Is 203 registered with freeswitch?
thanks Matt
On 06/06/2014 07:47, Richard Mace wrote:
Hi All, I have a vanilla debian 7 FreeSWITCH installation with a file called 201.xml in /etc/freeswitch/dialplan/default/
When I dial 201 (from extension 203) it won't ring the extension. I have included a SIP trace on: www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt
Any ideas what I'm doing wrong?
Thanks, as always, in advance
Richard
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
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you can use "sofia status profile internal" (without quotes) in the fs_cli. This will show you how many registrations you have. I think I was a bit hasty in my reply though, the call seems to be allowed from 203 but you are getting a "400 - Normal_temporary_failure" result when trying to connect to 201. This may be a networking issue due to nat settings (I see that 201 is using a 192 address). The sofia status command will show if you have auto-nat set or not but unfortunately I'm not too hot on nat traversal, so will have to leave the answer to a more qualified user on the mailing list. Hope you get it sorted :) Thanks Matt On 06/06/2014 10:57, Richard Mace wrote:
Hi Matt, I think so, what's the best command to check?
Richard
On 6 June 2014 09:42, Matt Broad <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
HI Richard,
it seems like your acl settings are blocking the incoming call. Is 203 registered with freeswitch?
thanks Matt
On 06/06/2014 07:47, Richard Mace wrote:
Hi All, I have a vanilla debian 7 FreeSWITCH installation with a file called 201.xml in /etc/freeswitch/dialplan/default/
When I dial 201 (from extension 203) it won't ring the extension. I have included a SIP trace on: www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt <http://www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt>
Any ideas what I'm doing wrong?
Thanks, as always, in advance
Richard
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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 <mailto: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 <mailto: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
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Thanks for your reply. Just to clarify, extension 203 can call the standard vanilla ivr on extension 5000 so I am wondering whether it is to do with the separate .XML file as the extension 5000 is in the default.XML file? Richard On 6 Jun 2014 11:39, "Matt Broad" <matt@inveroak.com> wrote:
you can use "sofia status profile internal" (without quotes) in the fs_cli. This will show you how many registrations you have.
I think I was a bit hasty in my reply though, the call seems to be allowed from 203 but you are getting a "400 - Normal_temporary_failure" result when trying to connect to 201. This may be a networking issue due to nat settings (I see that 201 is using a 192 address).
The sofia status command will show if you have auto-nat set or not but unfortunately I'm not too hot on nat traversal, so will have to leave the answer to a more qualified user on the mailing list.
Hope you get it sorted :)
Thanks Matt
On 06/06/2014 10:57, Richard Mace wrote:
Hi Matt, I think so, what's the best command to check?
Richard
On 6 June 2014 09:42, Matt Broad <matt@inveroak.com> wrote:
HI Richard,
it seems like your acl settings are blocking the incoming call. Is 203 registered with freeswitch?
thanks Matt
On 06/06/2014 07:47, Richard Mace wrote:
Hi All, I have a vanilla debian 7 FreeSWITCH installation with a file called 201.xml in /etc/freeswitch/dialplan/default/
When I dial 201 (from extension 203) it won't ring the extension. I have included a SIP trace on: www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt
Any ideas what I'm doing wrong?
Thanks, as always, in advance
Richard
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
No problem, I think the issue is more that ext 201 cannot be reached by freeswitch. You can test this by using the following in the fs_cli: *originate user/201 9198 XML Default* This will attempt to call 201 and execute extension 9198. If successful, you should get a call to 201 and be played the very catchy Tetris music. See how you get on with that. thanks Matt On 06/06/2014 12:26, Richard Mace wrote:
Thanks for your reply. Just to clarify, extension 203 can call the standard vanilla ivr on extension 5000 so I am wondering whether it is to do with the separate .XML file as the extension 5000 is in the default.XML file? Richard
On 6 Jun 2014 11:39, "Matt Broad" <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
you can use "sofia status profile internal" (without quotes) in the fs_cli. This will show you how many registrations you have.
I think I was a bit hasty in my reply though, the call seems to be allowed from 203 but you are getting a "400 - Normal_temporary_failure" result when trying to connect to 201. This may be a networking issue due to nat settings (I see that 201 is using a 192 address).
The sofia status command will show if you have auto-nat set or not but unfortunately I'm not too hot on nat traversal, so will have to leave the answer to a more qualified user on the mailing list.
Hope you get it sorted :)
Thanks Matt
On 06/06/2014 10:57, Richard Mace wrote:
Hi Matt, I think so, what's the best command to check?
Richard
On 6 June 2014 09:42, Matt Broad <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
HI Richard,
it seems like your acl settings are blocking the incoming call. Is 203 registered with freeswitch?
thanks Matt
On 06/06/2014 07:47, Richard Mace wrote:
Hi All, I have a vanilla debian 7 FreeSWITCH installation with a file called 201.xml in /etc/freeswitch/dialplan/default/
When I dial 201 (from extension 203) it won't ring the extension. I have included a SIP trace on: www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt <http://www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt>
Any ideas what I'm doing wrong?
Thanks, as always, in advance
Richard
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
Thanks Matt, I'll test that. I think that I should also mention that extension 201 can also successfully call ext 5000? Richard On 6 June 2014 13:28, Matt Broad <matt@inveroak.com> wrote:
No problem, I think the issue is more that ext 201 cannot be reached by freeswitch. You can test this by using the following in the fs_cli: *originate user/201 9198 XML Default* This will attempt to call 201 and execute extension 9198. If successful, you should get a call to 201 and be played the very catchy Tetris music. See how you get on with that.
thanks Matt
On 06/06/2014 12:26, Richard Mace wrote:
Thanks for your reply. Just to clarify, extension 203 can call the standard vanilla ivr on extension 5000 so I am wondering whether it is to do with the separate .XML file as the extension 5000 is in the default.XML file? Richard On 6 Jun 2014 11:39, "Matt Broad" <matt@inveroak.com> wrote:
you can use "sofia status profile internal" (without quotes) in the fs_cli. This will show you how many registrations you have.
I think I was a bit hasty in my reply though, the call seems to be allowed from 203 but you are getting a "400 - Normal_temporary_failure" result when trying to connect to 201. This may be a networking issue due to nat settings (I see that 201 is using a 192 address).
The sofia status command will show if you have auto-nat set or not but unfortunately I'm not too hot on nat traversal, so will have to leave the answer to a more qualified user on the mailing list.
Hope you get it sorted :)
Thanks Matt
On 06/06/2014 10:57, Richard Mace wrote:
Hi Matt, I think so, what's the best command to check?
Richard
On 6 June 2014 09:42, Matt Broad <matt@inveroak.com> wrote:
HI Richard,
it seems like your acl settings are blocking the incoming call. Is 203 registered with freeswitch?
thanks Matt
On 06/06/2014 07:47, Richard Mace wrote:
Hi All, I have a vanilla debian 7 FreeSWITCH installation with a file called 201.xml in /etc/freeswitch/dialplan/default/
When I dial 201 (from extension 203) it won't ring the extension. I have included a SIP trace on: www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt
Any ideas what I'm doing wrong?
Thanks, as always, in advance
Richard
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
ok, test the originate command and see what it comes back with. My guess would be a natting issue, and that Freeswitch is not able to connect to the phone using ext 201. thanks Matt On 06/06/2014 14:42, Richard Mace wrote:
Thanks Matt, I'll test that. I think that I should also mention that extension 201 can also successfully call ext 5000?
Richard
On 6 June 2014 13:28, Matt Broad <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
No problem, I think the issue is more that ext 201 cannot be reached by freeswitch. You can test this by using the following in the fs_cli: *originate user/201 9198 XML Default* This will attempt to call 201 and execute extension 9198. If successful, you should get a call to 201 and be played the very catchy Tetris music. See how you get on with that.
thanks Matt
On 06/06/2014 12:26, Richard Mace wrote:
Thanks for your reply. Just to clarify, extension 203 can call the standard vanilla ivr on extension 5000 so I am wondering whether it is to do with the separate .XML file as the extension 5000 is in the default.XML file? Richard
On 6 Jun 2014 11:39, "Matt Broad" <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
you can use "sofia status profile internal" (without quotes) in the fs_cli. This will show you how many registrations you have.
I think I was a bit hasty in my reply though, the call seems to be allowed from 203 but you are getting a "400 - Normal_temporary_failure" result when trying to connect to 201. This may be a networking issue due to nat settings (I see that 201 is using a 192 address).
The sofia status command will show if you have auto-nat set or not but unfortunately I'm not too hot on nat traversal, so will have to leave the answer to a more qualified user on the mailing list.
Hope you get it sorted :)
Thanks Matt
On 06/06/2014 10:57, Richard Mace wrote:
Hi Matt, I think so, what's the best command to check?
Richard
On 6 June 2014 09:42, Matt Broad <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
HI Richard,
it seems like your acl settings are blocking the incoming call. Is 203 registered with freeswitch?
thanks Matt
On 06/06/2014 07:47, Richard Mace wrote:
Hi All, I have a vanilla debian 7 FreeSWITCH installation with a file called 201.xml in /etc/freeswitch/dialplan/default/
When I dial 201 (from extension 203) it won't ring the extension. I have included a SIP trace on: www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt <http://www.rocksolidpbx.co.uk/download/sofia_sip_trace.txt>
Any ideas what I'm doing wrong?
Thanks, as always, in advance
Richard
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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OK, will do. ext 201 is on the same LAN as FreeSWITCH though? Richard On 6 June 2014 15:21, Matt Broad <matt@inveroak.com> wrote:
ok, test the originate command and see what it comes back with.
My guess would be a natting issue, and that Freeswitch is not able to connect to the phone using ext 201.
thanks Matt
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance. thanks Matt On 06/06/2014 15:46, Richard Mace wrote:
OK, will do. ext 201 is on the same LAN as FreeSWITCH though?
Richard
On 6 June 2014 15:21, Matt Broad <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
ok, test the originate command and see what it comes back with.
My guess would be a natting issue, and that Freeswitch is not able to connect to the phone using ext 201.
thanks Matt
_________________________________________________________________________ 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
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Thanks for your help Matt. Did I include a good enough sip trace in my first post? Richard On 6 June 2014 16:07, Matt Broad <matt@inveroak.com> wrote:
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance.
thanks Matt
Is this the vanilla config of Freeswitch? Vanilla only recognizes extensions 1000 to 1019, unless you modify the match string in the dialplan. -Francis On 7/06/2014 5:53 AM, Richard Mace wrote:
Thanks for your help Matt. Did I include a good enough sip trace in my first post?
Richard
On 6 June 2014 16:07, Matt Broad <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance.
thanks Matt
_________________________________________________________________________ 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
Hi Francis, It is a vanilla install, with the addition of extensions 201,202 and 203 all saved as seperate .xml files in /etc/freeswitch/dialplan/default/ directory Richard On 7 June 2014 10:57, Francis <sms@icefire.qza.net.au> wrote:
Is this the vanilla config of Freeswitch? Vanilla only recognizes extensions 1000 to 1019, unless you modify the match string in the dialplan.
-Francis
On 7/06/2014 5:53 AM, Richard Mace wrote:
Thanks for your help Matt. Did I include a good enough sip trace in my first post?
Richard
On 6 June 2014 16:07, Matt Broad <matt@inveroak.com> wrote:
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance.
thanks Matt
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
Have you modified the file: /etc/freeswitch/dialplan/default.xml to recognize this numbering plan? If not, just saving extensions won't work. -Francis On 7/06/2014 9:30 PM, Richard Mace wrote:
Hi Francis, It is a vanilla install, with the addition of extensions 201,202 and 203 all saved as seperate .xml files in /etc/freeswitch/dialplan/default/ directory
Richard
On 7 June 2014 10:57, Francis <sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>> wrote:
Is this the vanilla config of Freeswitch? Vanilla only recognizes extensions 1000 to 1019, unless you modify the match string in the dialplan.
-Francis
On 7/06/2014 5:53 AM, Richard Mace wrote:
Thanks for your help Matt. Did I include a good enough sip trace in my first post?
Richard
On 6 June 2014 16:07, Matt Broad <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance.
thanks Matt
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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 <mailto: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 <mailto: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
Hi Franics, When you say "recognize this numbering plan" do you mean this line?: <!-- You can place files in the default directory to get included. --> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> Or am I missing something else? Richard On 7 June 2014 12:45, Francis <sms@icefire.qza.net.au> wrote:
Have you modified the file:
/etc/freeswitch/dialplan/default.xml
to recognize this numbering plan? If not, just saving extensions won't work.
-Francis
On 7/06/2014 9:30 PM, Richard Mace wrote:
Hi Francis, It is a vanilla install, with the addition of extensions 201,202 and 203 all saved as seperate .xml files in /etc/freeswitch/dialplan/default/ directory
Richard
On 7 June 2014 10:57, Francis <sms@icefire.qza.net.au> wrote:
Is this the vanilla config of Freeswitch? Vanilla only recognizes extensions 1000 to 1019, unless you modify the match string in the dialplan.
-Francis
On 7/06/2014 5:53 AM, Richard Mace wrote:
Thanks for your help Matt. Did I include a good enough sip trace in my first post?
Richard
On 6 June 2014 16:07, Matt Broad <matt@inveroak.com> wrote:
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance.
thanks Matt
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
Close, that tells it to include the files you created, but it doesn't tell FS to actually recognize them when they are dialed. There is an extension in this file with a condition statement: <condition field="destination_number" expression="^(10[0,1][1-9])$"> You should see some comments before the extension describing what it is and does. This extension has all the actions that are/can be executed when an internal endpoint is dialed. As you can see, the statement above uses PCRE to match the number dialed, and it needs to match your desired numbering scheme in order to work. Just make sure that when you modify it, that it doesn't clash with any of the other extensions in the dialplan, most of which are in this file. If you haven't already, have a sift though the wiki on dialplan settings: https://wiki.freeswitch.org/wiki/Dialplan_XML Francis On 8/06/2014 2:42 AM, Richard Mace wrote:
Hi Franics, When you say "recognize this numbering plan" do you mean this line?:
<!-- You can place files in the default directory to get included. --> <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
Or am I missing something else?
Richard
On 7 June 2014 12:45, Francis <sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>> wrote:
Have you modified the file:
/etc/freeswitch/dialplan/default.xml
to recognize this numbering plan? If not, just saving extensions won't work.
-Francis
On 7/06/2014 9:30 PM, Richard Mace wrote:
Hi Francis, It is a vanilla install, with the addition of extensions 201,202 and 203 all saved as seperate .xml files in /etc/freeswitch/dialplan/default/ directory
Richard
On 7 June 2014 10:57, Francis <sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>> wrote:
Is this the vanilla config of Freeswitch? Vanilla only recognizes extensions 1000 to 1019, unless you modify the match string in the dialplan.
-Francis
On 7/06/2014 5:53 AM, Richard Mace wrote:
Thanks for your help Matt. Did I include a good enough sip trace in my first post?
Richard
On 6 June 2014 16:07, Matt Broad <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance.
thanks Matt
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
Hi Francis, I have actually commented that out, and replaced it with multiple files, in the /etc/freeswitch/dialplan/default/ such as this, which I believe is the same as splitting out the vanilla local extension, that is, when 203 is dialled, FreeSWITCH would look through all of the .xml files until it found this code, and would then ring the extension?: <include> <extension name="203"> <condition field="destination_number" expression="^(203)$"> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${uk-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=14"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> </include> Richard On 7 June 2014 18:23, Francis <sms@icefire.qza.net.au> wrote:
Close, that tells it to include the files you created, but it doesn't tell FS to actually recognize them when they are dialed. There is an extension in this file with a condition statement:
<condition field="destination_number" expression="^(10[0,1][1-9])$">
You should see some comments before the extension describing what it is and does. This extension has all the actions that are/can be executed when an internal endpoint is dialed. As you can see, the statement above uses PCRE to match the number dialed, and it needs to match your desired numbering scheme in order to work. Just make sure that when you modify it, that it doesn't clash with any of the other extensions in the dialplan, most of which are in this file. If you haven't already, have a sift though the wiki on dialplan settings:
https://wiki.freeswitch.org/wiki/Dialplan_XML
Francis
On 8/06/2014 2:42 AM, Richard Mace wrote:
Hi Franics, When you say "recognize this numbering plan" do you mean this line?:
<!-- You can place files in the default directory to get included. --> <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
Or am I missing something else?
Richard
On 7 June 2014 12:45, Francis <sms@icefire.qza.net.au> wrote:
Have you modified the file:
/etc/freeswitch/dialplan/default.xml
to recognize this numbering plan? If not, just saving extensions won't work.
-Francis
On 7/06/2014 9:30 PM, Richard Mace wrote:
Hi Francis, It is a vanilla install, with the addition of extensions 201,202 and 203 all saved as seperate .xml files in /etc/freeswitch/dialplan/default/ directory
Richard
On 7 June 2014 10:57, Francis <sms@icefire.qza.net.au> wrote:
Is this the vanilla config of Freeswitch? Vanilla only recognizes extensions 1000 to 1019, unless you modify the match string in the dialplan.
-Francis
On 7/06/2014 5:53 AM, Richard Mace wrote:
Thanks for your help Matt. Did I include a good enough sip trace in my first post?
Richard
On 6 June 2014 16:07, Matt Broad <matt@inveroak.com> wrote:
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance.
thanks Matt
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
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This would make sense in Asterisk, but FS differentiates between 'extensions' and 'users'. An extension is simply a destination for a number, with a set of conditions and actions. A user is still necessary if you want to register a phone or a soft client. Hence, FS vanilla only declares the extension once, but has multiple users that reference to it. Declaring the extension multiple times isn't necessary and has no effect since there is no user-no.xml to reference to it, so it fails. Simply changing the destination_number condition to match your number range and creating normal users (like the 1000 - 1019 that come with vanilla) is enough to do what you are trying to do. The Freeswitch 1.2 book on PaktPub is well worth the coin if you have it, just to save many hours of tinkering and being trapped by 'Asterisk logic'. FS has built in heuristics for detecting guesswork and it exacts the most terrifying revenge upon the offender. Francis On 8/06/2014 6:09 AM, Richard Mace wrote:
Hi Francis, I have actually commented that out, and replaced it with multiple files, in the /etc/freeswitch/dialplan/default/ such as this, which I believe is the same as splitting out the vanilla local extension, that is, when 203 is dialled, FreeSWITCH would look through all of the .xml files until it found this code, and would then ring the extension?:
<include> <extension name="203"> <condition field="destination_number" expression="^(203)$"> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${uk-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=14"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> </include>
Richard
On 7 June 2014 18:23, Francis <sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>> wrote:
Close, that tells it to include the files you created, but it doesn't tell FS to actually recognize them when they are dialed. There is an extension in this file with a condition statement:
<condition field="destination_number" expression="^(10[0,1][1-9])$">
You should see some comments before the extension describing what it is and does. This extension has all the actions that are/can be executed when an internal endpoint is dialed. As you can see, the statement above uses PCRE to match the number dialed, and it needs to match your desired numbering scheme in order to work. Just make sure that when you modify it, that it doesn't clash with any of the other extensions in the dialplan, most of which are in this file. If you haven't already, have a sift though the wiki on dialplan settings:
https://wiki.freeswitch.org/wiki/Dialplan_XML
Francis
On 8/06/2014 2:42 AM, Richard Mace wrote:
Hi Franics, When you say "recognize this numbering plan" do you mean this line?:
<!-- You can place files in the default directory to get included. --> <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
Or am I missing something else?
Richard
On 7 June 2014 12:45, Francis <sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>> wrote:
Have you modified the file:
/etc/freeswitch/dialplan/default.xml
to recognize this numbering plan? If not, just saving extensions won't work.
-Francis
On 7/06/2014 9:30 PM, Richard Mace wrote:
Hi Francis, It is a vanilla install, with the addition of extensions 201,202 and 203 all saved as seperate .xml files in /etc/freeswitch/dialplan/default/ directory
Richard
On 7 June 2014 10:57, Francis <sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>> wrote:
Is this the vanilla config of Freeswitch? Vanilla only recognizes extensions 1000 to 1019, unless you modify the match string in the dialplan.
-Francis
On 7/06/2014 5:53 AM, Richard Mace wrote:
Thanks for your help Matt. Did I include a good enough sip trace in my first post?
Richard
On 6 June 2014 16:07, Matt Broad <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance.
thanks Matt
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Hi Francis, Thanks for your detailed email. I have purchased the FS 1.2 book and read it a few times, I think I might have to go back and read it again :) The reason I wanted to do it in "per extension" files was because I am creating a FS GUI, and want to give users the choice, on a per extension basis, as to whether their extension diverts to voice mail or not, such as 200 and 202 may want to divert to voicemail, whereas 201 and 203 don't want that. So, I thought that if I had separate .xml files for each extension in the /etc/freeswitch/dialplan/default/ directory, then I could have any behaviour I wanted on a per "extension dialled" basis, unless of course I am completely missing something here? :) Richard On 8 June 2014 02:27, Francis <sms@icefire.qza.net.au> wrote:
This would make sense in Asterisk, but FS differentiates between 'extensions' and 'users'. An extension is simply a destination for a number, with a set of conditions and actions. A user is still necessary if you want to register a phone or a soft client. Hence, FS vanilla only declares the extension once, but has multiple users that reference to it. Declaring the extension multiple times isn't necessary and has no effect since there is no user-no.xml to reference to it, so it fails. Simply changing the destination_number condition to match your number range and creating normal users (like the 1000 - 1019 that come with vanilla) is enough to do what you are trying to do.
The Freeswitch 1.2 book on PaktPub is well worth the coin if you have it, just to save many hours of tinkering and being trapped by 'Asterisk logic'. FS has built in heuristics for detecting guesswork and it exacts the most terrifying revenge upon the offender.
Francis
On 8/06/2014 6:09 AM, Richard Mace wrote:
Hi Francis, I have actually commented that out, and replaced it with multiple files, in the /etc/freeswitch/dialplan/default/ such as this, which I believe is the same as splitting out the vanilla local extension, that is, when 203 is dialled, FreeSWITCH would look through all of the .xml files until it found this code, and would then ring the extension?:
<include> <extension name="203"> <condition field="destination_number" expression="^(203)$"> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${uk-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=14"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> </include>
Richard
On 7 June 2014 18:23, Francis <sms@icefire.qza.net.au> wrote:
Close, that tells it to include the files you created, but it doesn't tell FS to actually recognize them when they are dialed. There is an extension in this file with a condition statement:
<condition field="destination_number" expression="^(10[0,1][1-9])$">
You should see some comments before the extension describing what it is and does. This extension has all the actions that are/can be executed when an internal endpoint is dialed. As you can see, the statement above uses PCRE to match the number dialed, and it needs to match your desired numbering scheme in order to work. Just make sure that when you modify it, that it doesn't clash with any of the other extensions in the dialplan, most of which are in this file. If you haven't already, have a sift though the wiki on dialplan settings:
https://wiki.freeswitch.org/wiki/Dialplan_XML
Francis
On 8/06/2014 2:42 AM, Richard Mace wrote:
Hi Franics, When you say "recognize this numbering plan" do you mean this line?:
<!-- You can place files in the default directory to get included. --> <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
Or am I missing something else?
Richard
On 7 June 2014 12:45, Francis <sms@icefire.qza.net.au> wrote:
Have you modified the file:
/etc/freeswitch/dialplan/default.xml
to recognize this numbering plan? If not, just saving extensions won't work.
-Francis
On 7/06/2014 9:30 PM, Richard Mace wrote:
Hi Francis, It is a vanilla install, with the addition of extensions 201,202 and 203 all saved as seperate .xml files in /etc/freeswitch/dialplan/default/ directory
Richard
On 7 June 2014 10:57, Francis <sms@icefire.qza.net.au> wrote:
Is this the vanilla config of Freeswitch? Vanilla only recognizes extensions 1000 to 1019, unless you modify the match string in the dialplan.
-Francis
On 7/06/2014 5:53 AM, Richard Mace wrote:
Thanks for your help Matt. Did I include a good enough sip trace in my first post?
Richard
On 6 June 2014 16:07, Matt Broad <matt@inveroak.com> wrote:
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance.
thanks Matt
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
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Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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Ah I see! There is an option somewhere on the wiki (memory fails me right now) to disable voicemail on a per user basis, set in their profile. In the case of setting variables per user, it's best to keep them in the user profile so that they are called when the user makes a call. In the case of pulling variables when the user is _being_ called, you can execute actions inline to pull and set those variables when the call hits the extension in the dialplan. I've done this on intercom calls, where you can set variables in both the caller's and callee's profile, then the extension just sets them when a call is made. The idea is that you only create extensions for different functions just once, and keep user settings with the users. There are some exceptions, like when you need some convoluted ring group strategy and using the groups feature is too limited to accomodate this, so it's better to list the users in an extension that does what you want. Another case would be when you have a group of extensions that you want to behave fundamentally different from another group, then it might make sense to create a different extension and match it to a specific range. There are two places that matter when it comes to users: /etc/freeswitch/dialplan/default.xml /etc/freeswitch/directory/default/<user-profiles-go-here.xml> The dialplan is for dialling/execution, nothing more or less. Extensions are not users or endpoints, they are just chunks of code that do things. The directory is for users/endpoints. You need both for a call to work, otherwise all those xml extensions are trying to call users that don't exist, and FS expects them to be there. If you wanted your gui to use multiple extensions, you'd also have to make it create a user profile as well for each user. Again, this is going to add a very large amount of redundency to your system if it is creating the same core functions over and over, when they could be just declared once and matched to a group of users. It also opens your system up to a performance penalty, because FS has to traverse the dialplan until it finds a match, and if you have 200 users each with a unique extension, then that's 200 extensions hogging memory and up to 200 attempts that FS has to try before it finds a match, whereas having only one extension means that it only has one extension to load and match. The ideal situation is to build the functionality into the dialplan that you want, catering for all scenarios that you are expecting to encounter, then use variables in the user's profiles to trigger those functions as needed. Francis On 8/06/2014 4:04 PM, Richard Mace wrote:
Hi Francis, Thanks for your detailed email. I have purchased the FS 1.2 book and read it a few times, I think I might have to go back and read it again :) The reason I wanted to do it in "per extension" files was because I am creating a FS GUI, and want to give users the choice, on a per extension basis, as to whether their extension diverts to voice mail or not, such as 200 and 202 may want to divert to voicemail, whereas 201 and 203 don't want that. So, I thought that if I had separate .xml files for each extension in the /etc/freeswitch/dialplan/default/ directory, then I could have any behaviour I wanted on a per "extension dialled" basis, unless of course I am completely missing something here? :)
Richard
On 8 June 2014 02:27, Francis <sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>> wrote:
This would make sense in Asterisk, but FS differentiates between 'extensions' and 'users'. An extension is simply a destination for a number, with a set of conditions and actions. A user is still necessary if you want to register a phone or a soft client. Hence, FS vanilla only declares the extension once, but has multiple users that reference to it. Declaring the extension multiple times isn't necessary and has no effect since there is no user-no.xml to reference to it, so it fails. Simply changing the destination_number condition to match your number range and creating normal users (like the 1000 - 1019 that come with vanilla) is enough to do what you are trying to do.
The Freeswitch 1.2 book on PaktPub is well worth the coin if you have it, just to save many hours of tinkering and being trapped by 'Asterisk logic'. FS has built in heuristics for detecting guesswork and it exacts the most terrifying revenge upon the offender.
Francis
On 8/06/2014 6:09 AM, Richard Mace wrote:
Hi Francis, I have actually commented that out, and replaced it with multiple files, in the /etc/freeswitch/dialplan/default/ such as this, which I believe is the same as splitting out the vanilla local extension, that is, when 203 is dialled, FreeSWITCH would look through all of the .xml files until it found this code, and would then ring the extension?:
<include> <extension name="203"> <condition field="destination_number" expression="^(203)$"> <action application="export" data="dialed_extension=$1"/> <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app> --> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${uk-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=14"/> <!-- <action application="set" data="sip_exclude_contact=${network_addr}"/> --> <action application="set" data="hangup_after_bridge=true"/> <!--<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/> --> <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>--> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> </include>
Richard
On 7 June 2014 18:23, Francis <sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>> wrote:
Close, that tells it to include the files you created, but it doesn't tell FS to actually recognize them when they are dialed. There is an extension in this file with a condition statement:
<condition field="destination_number" expression="^(10[0,1][1-9])$">
You should see some comments before the extension describing what it is and does. This extension has all the actions that are/can be executed when an internal endpoint is dialed. As you can see, the statement above uses PCRE to match the number dialed, and it needs to match your desired numbering scheme in order to work. Just make sure that when you modify it, that it doesn't clash with any of the other extensions in the dialplan, most of which are in this file. If you haven't already, have a sift though the wiki on dialplan settings:
https://wiki.freeswitch.org/wiki/Dialplan_XML
Francis
On 8/06/2014 2:42 AM, Richard Mace wrote:
Hi Franics, When you say "recognize this numbering plan" do you mean this line?:
<!-- You can place files in the default directory to get included. --> <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
Or am I missing something else?
Richard
On 7 June 2014 12:45, Francis <sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>> wrote:
Have you modified the file:
/etc/freeswitch/dialplan/default.xml
to recognize this numbering plan? If not, just saving extensions won't work.
-Francis
On 7/06/2014 9:30 PM, Richard Mace wrote:
Hi Francis, It is a vanilla install, with the addition of extensions 201,202 and 203 all saved as seperate .xml files in /etc/freeswitch/dialplan/default/ directory
Richard
On 7 June 2014 10:57, Francis <sms@icefire.qza.net.au <mailto:sms@icefire.qza.net.au>> wrote:
Is this the vanilla config of Freeswitch? Vanilla only recognizes extensions 1000 to 1019, unless you modify the match string in the dialplan.
-Francis
On 7/06/2014 5:53 AM, Richard Mace wrote:
Thanks for your help Matt. Did I include a good enough sip trace in my first post?
Richard
On 6 June 2014 16:07, Matt Broad <matt@inveroak.com <mailto:matt@inveroak.com>> wrote:
strange :| Sorry to say but I think I may have to step out and let someone else see if they can offer some assistance.
thanks Matt
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Hi Francis, Many thanks for taking the time to explain this so well for me. I hadn't appreciated the design of FS like this before, and now it make alot more sense. I'll make some adjustments to the way my GUI works now, and see how things pan out. :) Richard On 8 June 2014 15:29, Francis <sms@icefire.qza.net.au> wrote:
Ah I see! There is an option somewhere on the wiki (memory fails me right now) to disable voicemail on a per user basis, set in their profile. In the case of setting variables per user, it's best to keep them in the user profile so that they are called when the user makes a call. In the case of pulling variables when the user is _being_ called, you can execute actions inline to pull and set those variables when the call hits the extension in the dialplan. I've done this on intercom calls, where you can set variables in both the caller's and callee's profile, then the extension just sets them when a call is made. The idea is that you only create extensions for different functions just once, and keep user settings with the users. There are some exceptions, like when you need some convoluted ring group strategy and using the groups feature is too limited to accomodate this, so it's better to list the users in an extension that does what you want. Another case would be when you have a group of extensions that you want to behave fundamentally different from another group, then it might make sense to create a different extension and match it to a specific range. There are two places that matter when it comes to users:
/etc/freeswitch/dialplan/default.xml /etc/freeswitch/directory/default/<user-profiles-go-here.xml>
The dialplan is for dialling/execution, nothing more or less. Extensions are not users or endpoints, they are just chunks of code that do things. The directory is for users/endpoints. You need both for a call to work, otherwise all those xml extensions are trying to call users that don't exist, and FS expects them to be there. If you wanted your gui to use multiple extensions, you'd also have to make it create a user profile as well for each user. Again, this is going to add a very large amount of redundency to your system if it is creating the same core functions over and over, when they could be just declared once and matched to a group of users. It also opens your system up to a performance penalty, because FS has to traverse the dialplan until it finds a match, and if you have 200 users each with a unique extension, then that's 200 extensions hogging memory and up to 200 attempts that FS has to try before it finds a match, whereas having only one extension means that it only has one extension to load and match. The ideal situation is to build the functionality into the dialplan that you want, catering for all scenarios that you are expecting to encounter, then use variables in the user's profiles to trigger those functions as needed.
Francis
On 8/06/2014 4:04 PM, Richard Mace wrote:
Hi Francis, Thanks for your detailed email. I have purchased the FS 1.2 book and read it a few times, I think I might have to go back and read it again :) The reason I wanted to do it in "per extension" files was because I am creating a FS GUI, and want to give users the choice, on a per extension basis, as to whether their extension diverts to voice mail or not, such as 200 and 202 may want to divert to voicemail, whereas 201 and 203 don't want that. So, I thought that if I had separate .xml files for each extension in the /etc/freeswitch/dialplan/default/ directory, then I could have any behaviour I wanted on a per "extension dialled" basis, unless of course I am completely missing something here? :)
Richard
On 8 June 2014 02:27, Francis <sms@icefire.qza.net.au> wrote:
This would make sense in Asterisk, but FS differentiates between 'extensions' and 'users'. An extension is simply a destination for a number, with a set of conditions and actions. A user is still necessary if you want to register a phone or a soft client. Hence, FS vanilla only declares the extension once, but has multiple users that reference to it. Declaring the extension multiple times isn't necessary and has no effect since there is no user-no.xml to reference to it, so it fails. Simply changing the destination_number condition to match your number range and creating normal users (like the 1000 - 1019 that come with vanilla) is enough to do what you are trying to do.
The Freeswitch 1.2 book on PaktPub is well worth the coin if you have it, just to save many hours of tinkering and being trapped by 'Asterisk logic'. FS has built in heuristics for detecting guesswork and it exacts the most terrifying revenge upon the offender.
Francis
On Mon, Jun 9, 2014 at 3:43 AM, Richard Mace <richard.mace@gmail.com> wrote:
Hi Francis, Many thanks for taking the time to explain this so well for me. I hadn't appreciated the design of FS like this before, and now it make alot more sense. I'll make some adjustments to the way my GUI works now, and see how things pan out. :)
Richard
Another option, when you're ready to *really* ramp it up, is to do dynamic configs. Once powerful tool for that is mod_xml_curl: https://confluence.freeswit.ch/display/FREESWITCH/mod_xml_curl See also chapter 9 of the FreeSWITCH 1.2 book... -MC
Hi Michael. After reading that chapter, I was under the impression that using the XML configuration files would be better than the mod curl PBX config type work? Richard On 9 Jun 2014 18:26, "Michael Collins" <msc@freeswitch.org> wrote:
On Mon, Jun 9, 2014 at 3:43 AM, Richard Mace <richard.mace@gmail.com> wrote:
Hi Francis, Many thanks for taking the time to explain this so well for me. I hadn't appreciated the design of FS like this before, and now it make alot more sense. I'll make some adjustments to the way my GUI works now, and see how things pan out. :)
Richard
Another option, when you're ready to *really* ramp it up, is to do dynamic configs. Once powerful tool for that is mod_xml_curl: https://confluence.freeswit.ch/display/FREESWITCH/mod_xml_curl
See also chapter 9 of the FreeSWITCH 1.2 book...
-MC
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On Mon, Jun 9, 2014 at 11:06 AM, Richard Mace <richard.mace@gmail.com> wrote:
Hi Michael. After reading that chapter, I was under the impression that using the XML configuration files would be better than the mod curl PBX config type work? Richard
<http://www.freeswitch.org>I suppose it depends on what you're hoping to accomplish. I would imagine that in building a FS GUI the process of dynamically generating static-ish XML files would become unwieldy. Of course, if you're adept at wielding XML files then more power to you! -MC
participants (4)
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Francis
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Matt Broad
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Michael Collins
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Richard Mace