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 <[email protected]> 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 a...@default dial > {sip_auth_username=myuser,sip_auth_password=mypassword}sofia/external/ > [email protected] > > 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 a...@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 > > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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