(mad scientist laugh)!!! its alive!!!! thnx man ..... it WAS an issue w/ the prefix stripping in sipX. I used the default local dial plan. It specifies 9 as the prefix to call out to pstn w/c it strips before sending to gateway and external number length= 7 digits....so when i dial out 91234567....it only sends out 1234567 to the gateway and no match is found based on my dial peers....now the only problem is with the auto attendant not working....will post a new topic for that....thnx again....btw....no such thing as too much beer....unless ur already barfing while sleeping =P
-ron On Tue, Mar 30, 2010 at 4:43 AM, Staffan Kerker <[email protected]>wrote: > 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/ >
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