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 On Sun, Aug 2, 2009 at 4:36 PM, Adam Wilt <wiltingtree@gmail.com> wrote:

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@1.2.3.4

where 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/701

Both 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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