Hi Alberto, I'm wondering if my image might be the problem - I have 12.3.9 on the device - released at some point in may of this year. I've got everything (including the kitchen sink) in terms of feature set. Can you post some of the relevant snippets of your config? I'd love to see how this is done.
Graeme On Fri, 9 Jul 2004, Alberto Fernandez wrote: > I have an mc3800 working in my office with asterisk, you need the latest > vertion of ios. i have the image if you want it. Sip has a lot of bugs > on 12.2, > > I KNOW i went through hell > > > On Fri, 2004-07-09 at 09:20, [EMAIL PROTECTED] wrote: > > Hi Everyone, > > > > I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm > > wondering in anyone has got one of these suckers to work with asterisk in > > such a way that each FXS port has it's own extension. > > > > It speaks SIP, and I can send calls from asterisk out to it, but can't > > figure out how to get it to pass username & pw to asterisk when I try to > > configure it as a client. Eg - > > > > Call from a Grandstream (working)- > > > > Jul 8 20:53:57 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: > > Contact hop: <sip:[EMAIL PROTECTED]> > > -- Executing NoOp("SIP/4000-98ec", "") in new stack > > -- Executing Goto("SIP/4000-98ec", "intern-post|4001|1") in new stack > > -- Goto (intern-post,4001,1) > > -- Executing Dial("SIP/4000-98ec", "SIP/4001|30|Ttm") in new stack > > Jul 8 20:53:57 DEBUG[409616]: app_dial.c:420 dial_exec: SIMPLE DIAL (NO > > URL) > > Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:835 create_addr: Setting NAT on > > RTP to 0 > > Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1040 sip_call: Outgoing Call for > > 4001 > > Jul 8 20:53:57 DEBUG[409616]: chan_sip.c:1139 find_user: 4001 is not a > > local user > > -- Called 4001 > > Jul 8 20:53:57 DEBUG[409616]: channel.c:1402 ast_prod: Prodding channel > > 'SIP/4000-98ec' > > > > Call from the Cisco (not working) > > > > Jul 8 20:54:50 DEBUG[229391]: chan_sip.c:3643 build_route: build_route: > > Contact hop: <sip:[EMAIL PROTECTED]:5060> > > -- Executing NoOp("SIP/192.168.1.9-08134bb8", "") in new stack > > -- Executing Goto("SIP/192.168.1.9-08134bb8", "from-sip-post|4001|1") > > in new stack > > -- Goto (from-sip-post,4001,1) > > Jul 8 20:54:50 WARNING[426000]: pbx.c:1802 ast_pbx_run: Channel > > 'SIP/192.168.1.9-08134bb8' sent into invalid extension '4001' in context > > 'from-sip-post', but no invalid handler > > > > BTW- Working with a ripped-off version of John Todd's configs... Anyone > > get this working? It's kicking my ass. > > > > Jim > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users