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
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