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mrjoli021
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Integration with Freeswitch

Nov 12, 2013 4:41 PM

I am attempting to send certain calls to a freeswitch server.  I keep getting the following error

"error: Cannot make new call leg, request dropped as legService is not ready or is blocked." In the admin console

here are snapshots of my sip.xml and workflow.xml

[code]

<Profile>

                              <!-- Unique name for the profile. This ID is used in workflow to identify where the call will be placed -->

                              <profileID> FreeSwitch </profileID>

                              <!-- SIP username -->

                              <userName> 199 </userName>

                              <!-- SIP password -->

                              <password> 199 </password>

                              <!-- SIP display Name, can be same as username -->

                              <displayName> AMG </displayName>

                              <!-- Here the client is binded dynamically, in which case the sip phone's ip address is bounded by the register request -->

                              <remoteSipHost> freeswitchIP:5060 </remoteSipHost>

                              <!-- Global Address if the AMG server is on a local network and need to connect to outside network -->

                              <!-- <globalAddress> * </globalAddress> -->

                              <!-- Context in which incoming calls will be handled, this points to the context in workflow.xml -->

                              <context> rtmp </context>

                              <!-- supported Codecs for this SIP profile

                              All the codecs here should be from the CodecList defined earlier,

                              If no codec is specified then codecs specified in CodecList would be used

                              DTMF is always supported -->

                              <supportedCodecs>

                                        <codecID> G711u </codecID>

                              </supportedCodecs>

                    </Profile>

 

 

<Context name="sipPhoneContext">

          <!-- start conference with FreeSwitch -->

          <Condition variable="destNum" value="^433$">

                    <AppNode sequence="1" app="bridge" args="sip|${199}@freeswitchIP"/>

                    <AppNode sequence="2" app="hangup" args="null"/>

          </Condition>

 

[/code]

 

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