[Asterisk-Users] Cisco MC3810 -> Asterisk
jlaing at freaksh0.net
jlaing at freaksh0.net
Fri Jul 9 12:42:01 MST 2004
Thanks Alberto! I'm going to plug away with some of this and see how
things go for me. I'll post any differences required to get the FXS ports
going up.
Thanks again!
On Fri, 9 Jul 2004, Alberto Fernandez wrote:
> 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
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
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> > >
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