Adam,
Pastebin the logs. Also, a sip dump of both situations can really help.
To enable sip traces on FreeSWITCH all you have to do is type on the CLI:
sofia profile <profile> siptrace on/off
jmesquita
Hello,I'm trying to conference-in a call from FreeSWITCH to an extension on another PBX using sip.According to the documentation, I think it should look like this:conference abc@default dial {sip_auth_username=myuser,sip_auth_password=mypassword}sofia/external/101@1.2.3.4where 1.2.3.4 is the ip address of the remote pbx, and 101 is the extension.I've tried adding a gateway for it in the sip profiles, and then doing this:conference abc@default dial sofia/mygateway/701Both of these methods give me a result of:Call Requested: result: [DESTINATION_OUT_OF_ORDER]I set-up my soft phone to register to the same ip address with the same credentials, and it allows me to call the extension properly.Can somebody please tell me what I'm doing wrong?Thanks,Adam
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