Newb question on dialplan config
Hello, I am a FreeSwitch newb but have been using asterisk for a while now. I have a project for which I think FreeSwitch will be the best answer, so I need to learn. Have been reading the docs and followed the example at: http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk when I call from a Polycom on the asterisk box to a polycom on the freeswitch box all is good. When id do the reverse I.E. call the ast polycom from the freeswitch polycom I get only the following in the freswitch CLI: 2008-10-10 13:33:24 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/1002@192.168.100.16 [521c96a2-5205-bf46-9f9f-31124757b0ef] 2008-10-10 13:33:24 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing John Millican->2002 in context default 2008-10-10 13:33:24 [NOTICE] switch_ivr.c:1098 switch_ivr_session_transfer() Transfer sofia/internal/1002@192.168.100.16 to enum[2002@default] 2008-10-10 13:33:24 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-10-10 13:33:24 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/1002@192.168.100.16 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-10-10 13:33:24 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 12 (sofia/internal/1002@192.168.100.16) Ended 2008-10-10 13:33:24 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/1002@192.168.100.16 [CS_HANGUP] It would seem that the line: 2008-10-10 13:33:24 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting is telling me my problem but I do not yet know why freeswitch does not have a route. I am certain that I have not correctly set the dial plan but haven't a clue what to look at. Both machines are on the 192.168.100.0 net, firewall is off on both the freeswitch box which is running on a VMware installation of WinXP SP3 and the asterisk box. I am using the default configs with the additions per the above page. I did have to change the following from the defaults in vars.xml to get 2 way audio when I call from asterisk to freeswitch: <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.100.16"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=192.168.100.16"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=192.168.100.16"/> Any ideas? Is there something else I need to post to help decipher what I have done wrong or have not yet done? Thanks, JohnM
Its looking for extension 2002 in context default on FreeSWITCH and one doesn't exist so you get the NO ROUTE message. Add a route to map 2002 so that it points at the Asterisk box. /b On Oct 10, 2008, at 1:00 PM, John Millican wrote:
Processing John Millican->2002 in context defau
Brian West wrote:
Its looking for extension 2002 in context default on FreeSWITCH and one doesn't exist so you get the NO ROUTE message.
Add a route to map 2002 so that it points at the Asterisk box.
/b
On Oct 10, 2008, at 1:00 PM, John Millican wrote:
Processing John Millican->2002 in context defau I currently have this in default.xml in the context default: <extension name="ast_extens"> <condition field="destination_number" expression="^(2\d{3})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$1@192.168.100.11"/> <action application="hangup"/> </condition> </extension>
Is this not a routemap? I apologize for such simple questions, but I am learning. JohnM
I'm going to guess you added it at the very bottom of the default.xml? It needs to be above this line: <X-PRE-PROCESS cmd="include" data="default/*.xml"/> /b On Oct 10, 2008, at 1:22 PM, John Millican wrote:
I currently have this in default.xml in the context default: <extension name="ast_extens"> <condition field="destination_number" expression="^(2\d{3})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/ $1@192.168.100.11"/> <action application="hangup"/> </condition> </extension>
Is this not a routemap? I apologize for such simple questions, but I am learning. JohnM
Yep that was it! Now I just need to add a matching gateway, as I am getting the error no matching gateway found, which I "think" I can figure out. Thank you for such quick and accurate help. JohnM Brian West wrote:
I'm going to guess you added it at the very bottom of the default.xml?
It needs to be above this line:
<X-PRE-PROCESS cmd="include" data="default/*.xml"/>
/b
On Oct 10, 2008, at 1:22 PM, John Millican wrote:
I currently have this in default.xml in the context default: <extension name="ast_extens"> <condition field="destination_number" expression="^(2\d{3})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/external/$1@192.168.100.11 <mailto:sofia/external/$1@192.168.100.11>"/> <action application="hangup"/> </condition> </extension>
Is this not a routemap? I apologize for such simple questions, but I am learning. JohnM
The new default configs in SVN trunk have a HUGE warning at the very bottom along with more documentation. I highly recommend you check it out. http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/dialplan/default.xml?r=... /b On Oct 10, 2008, at 1:46 PM, John Millican wrote:
Yep that was it! Now I just need to add a matching gateway, as I am getting the error no matching gateway found, which I "think" I can figure out. Thank you for such quick and accurate help. JohnM
Thanks, will do that right now. JohnM Brian West wrote:
The new default configs in SVN trunk have a HUGE warning at the very bottom along with more documentation. I highly recommend you check it out.
http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/dialplan/default.xml?r=...
/b
On Oct 10, 2008, at 1:46 PM, John Millican wrote:
Yep that was it! Now I just need to add a matching gateway, as I am getting the error no matching gateway found, which I "think" I can figure out. Thank you for such quick and accurate help. JohnM
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Brian West
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John Millican