Adam,<br><br>Pastebin the logs. Also, a sip dump of both situations can really help.<br><br>To enable sip traces on FreeSWITCH all you have to do is type on the CLI:<br><br>sofia profile <profile> siptrace on/off<br>
<br><br>jmesquita<br><br><div class="gmail_quote">On Sun, Aug 2, 2009 at 4:36 PM, Adam Wilt <span dir="ltr"><<a href="mailto:wiltingtree@gmail.com">wiltingtree@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello,<div><br></div><div>I'm trying to conference-in a call from FreeSWITCH to an extension on another PBX using sip.</div><div><br></div><div>According to the documentation, I think it should look like this:</div><div>
<br></div><div> conference abc@default dial {sip_auth_username=myuser,sip_auth_password=mypassword}sofia/external/<a href="mailto:101@1.2.3.4" target="_blank">101@1.2.3.4</a></div><div><br></div><div>where 1.2.3.4 is the ip address of the remote pbx, and 101 is the extension.</div>
<div><br></div><div>I've tried adding a gateway for it in the sip profiles, and then doing this:</div><div><br></div><div> conference abc@default dial sofia/mygateway/701</div><div><br></div><div>Both of these methods give me a result of: </div>
<div><br></div><div> Call Requested: result: [DESTINATION_OUT_OF_ORDER]</div><div><br></div><div>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.</div>
<div><br></div><div>Can somebody please tell me what I'm doing wrong?</div><div><br></div><div>Thanks,</div><div>Adam</div><font color="#888888"><div><br></div>
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