[Asterisk-Users] Cisco MC3810 -> Asterisk

Alberto Fernandez asterisk at xynergia.net
Fri Jul 9 11:09:42 MST 2004 

Make sure that that the gatekeeper is turned off. these boys do both
gateways and gatekeeper...

Here is my CONF

Escape character is '^]'.
 
 
User Access Verification
 
Password:
3800>ena
Password:
3800#wr t
Building configuration...
 
Current configuration : 4796 bytes
!
version 12.3
no service pad
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname 3800
!

!
clock timezone GMT 0
network-clock base-rate 56k
ip subnet-zero
!
!
!
isdn switch-type primary-dms100
isdn voice-call-failure 3
!
voice hunt user-busy
voice call send-alert
voice call convert-discpi-to-prog
voice rtp send-recv
!
voice service voip
!
!
!
voice class h323 1
 h225 timeout tcp establish 3
!
!
voice class h323 2
  call start fast
!
!
!
!
!
!
no voice confirmation-tone
no voice hpi capture buffer
no voice hpi capture destination
!
!
!
!
!
controller T1 1
 framing esf
 linecode b8zs
 pri-group timeslots 1-24
translation-rule 99
 Rule 1 2604 099421549
!
!
!
!
interface Tunnel1
 no ip address
!
interface Ethernet0
 ip address MYROUTERIP 255.255.255.192
 no ip route-cache
 no ip mroute-cache
!
interface Serial0
 no ip address
 no ip route-cache
 no ip mroute-cache
 shutdown
!
interface Serial1
 no ip address
 no ip route-cache
 no ip mroute-cache
 shutdown
!
interface Serial1:23
 no ip address
 ip mroute-cache
 no logging event link-status
 isdn switch-type primary-dms100
 isdn incoming-voice modem 64
 isdn guard-timer 3000
 isdn map address .* plan unknown type unknown
 isdn T203 400000
 isdn T306 400000
 isdn T310 400000
 isdn send-alerting
 isdn negotiate-bchan
 isdn sending-complete
 keepalive 20
 no fair-queue
 no cdp enable
!
interface FR-ATM20
 no ip address
 shutdown
!
ip classless
ip route 0.0.0.0 0.0.0.0 DEFAULTGW
no ip http server
!
!
dialer-list 1 protocol ip permit
dialer-list 1 protocol ipx permit
!
!
!
voice-port 1:23
 !
 !
 dial-peer cor custom
!
!
!
dial-peer voice 1 pots
 incoming called-number .
 direct-inward-dial
!
dial-peer voice 7862 voip
 destination-pattern 2604
 progress_ind progress enable 8
 translate-outgoing called 99
 session protocol sipv2
 session target ipv4:IP OF SIPSERVER
 fax rate 14400
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
 no vad
!
dial-peer voice 305 pots
 destination-pattern 305.......
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 no digit-strip
 direct-inward-dial
 port 1:23
!
dial-peer voice 954 pots
 destination-pattern 954.......
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 1:23
 prefix 954
!
dial-peer voice 561 pots
 destination-pattern 561.......
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 1:23
 prefix 561
!
dial-peer voice 786 pots
 destination-pattern 786.......
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 1:23
 prefix 786
!
dial-peer voice 18 pots
 destination-pattern 18.........
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 1:23
 prefix 18
!
dial-peer voice 12 pots
 destination-pattern 1..........
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 1:23
 prefix 1
!
dial-peer voice 411 pots
 destination-pattern 411
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 1:23
 prefix 411
!
dial-peer voice 911 pots
 destination-pattern 911
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 1:23
 prefix 911
!
dial-peer voice 11 pots
 destination-pattern 011T
 progress_ind setup enable 3
 progress_ind progress enable 8
 progress_ind connect enable 8
 port 1:23
 prefix 011
!
dial-peer voice 7863 voip
 max-conn 2
 destination-pattern 440[4-5]
 progress_ind progress enable 8
 session protocol sipv2
 session target ipv4:IP OF SIPSERVER
 codec g711ulaw
 fax rate 14400
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
 no vad
!
dial-peer voice 20000 voip
 destination-pattern 2[6-8][0-9][0-9]
 progress_ind progress enable 8
 session protocol sipv2
 session target ipv4:IP OF SIPSERVER
 codec g711ulaw
 fax rate 14400
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
 no vad
!
gateway
!
sip-ua
 nat symmetric check-media-src
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 sip-server ipv4:IP OF SIPSERVER
!
!
gatekeeper
 shutdown
!
alias exec h sh isdn history
alias exec c sh call his voi bri | include Originate
alias exec sch sh call his voi bri
alias exec sca sh call ac voi bri
alias exec dial sh dial-peer voice sum
alias exec ca sh call ac voi bri | include Originate
alias exec ctc sh controllers t1 call-counters | inc DS0
alias exec a sh call ac voi bri
!
line con 0
line aux 0
line 2 3
 flush-at-activation
line vty 0 4
 login
!
!
end


On Fri, 2004-07-09 at 13:06, jlaing at freaksh0.net wrote:
> 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, jlaing at freaksh0.net 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:4000 at 192.168.1.42>
> > >     -- 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:4002 at 192.168.1.9:5060>
> > >     -- 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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> > > Asterisk-Users at lists.digium.com
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> > >
> > >
> >
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> 
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