On 29 mar 2010, at 12.04, ronald teng wrote: > I tried applying the changes but still cant make outgoing calls. btw, isn't > sip trunking used only if i have an e1/t1 line to use it with? i online have > a normals pots line....should i still enable it? Also, the auto attendants > seems to have stopped working....any ideas? posted both debug results and > updated config.....any further assistance would be greatly appreciated (if > only i can send u beer via email.... :)
Now, maybe I don't need any (more) beer 'cause I think i messed this dial-peer up... > dial-peer voice 21 pots > description *** Outbound to PSTN using FXO *** > incoming called-number 9.T > port 1/1/0 > ! It's an outgoing dial-peer so I think you should have "destination-pattern 9.T" instead of the "incoming called-number 9.T"... Try to change it and see if it does the trick. You cal also, just to check, enter the specific number you're dialing (98876005) to see if the router matches that one. Just to check, since I can't see the initial INVITE in your debug trace, is the SIP INVITE Request-URI actually showing the correct number of digits and prefixed by '9'? Just to make sure that you don't strip the '9' in SipX already... Your debug is clearly showing that the router can't find a matching dial-peer (404 Not Found) so it's all about that. Then, make sure that you run a "fresh" IOS, 'cause a lot of SIP fixes are in there in the latest releases. I will try to put a Cisco FXO and FXS example on the Wiki as well. Regards, /Staffan -- Staffan Kerker mail/sip/xmpp: [email protected] "There is absolutely no money above the 5th fret..." /Donald "Duck" Dunn _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
