09-23-2002 02:12 PM - edited 03-12-2019 08:54 PM
I currently have a 3640 w/ 12.2(11)T that is acting as a remote site SRST gateway. It is talking to a central 3.2 Call Manager. I configured the 3640 to be a MGCP gateway, with SRST and MGCP Fallback.
I am running into an issue w/ the FXS interface on the 3640. I have an analog phone connected that is able to place and receive calls while the 3640 is running in MGCP mode. When the WAN fails and H.323/SRST kicks in, the phone can still receive calls but is unable to place them. It gets a fast busy signal.
Here is the 3640 config:
version 12.2
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname XXXX
!
voice-card 1
!
voice-card 2
!
ip subnet-zero
!
!
!
!
!
!
voice call carrier capacity active
!
!
!
!
!
!
!
!
!
mta receive maximum-recipients 0
ccm-manager fallback-mgcp
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server X.X.X.X
ccm-manager config
!
controller T1 2/0
framing sf
linecode ami
!
controller T1 2/1
framing sf
linecode ami
!
!
!
!
interface FastEthernet0/0
no ip address
duplex auto
speed auto
!
interface FastEthernet0/0.1
encapsulation dot1Q 1 native
ip address X.X.X.X 255.255.254.0
!
interface FastEthernet0/0.250
encapsulation dot1Q 250
ip address X.X.X.X 255.255.255.0
!
interface Serial0/0
no ip address
encapsulation frame-relay
frame-relay class XXXX
frame-relay traffic-shaping
frame-relay lmi-type ansi
!
interface Serial0/0.101 point-to-point
bandwidth 384
ip address 192.168.10.1 255.255.255.0
frame-relay interface-dlci 101
!
router eigrp 100
network 10.0.0.0
network X.X.X.X
network 192.168.10.0
no auto-summary
no eigrp log-neighbor-changes
!
ip classless
ip http server
ip pim bidir-enable
!
!
!
map-class frame-relay XXX
frame-relay cir 384000
frame-relay bc 1000
frame-relay be 0
frame-relay mincir 384000
service-policy output voice-policy
frame-relay fragment 320
access-list 101 permit ip any host X.X.X.X
access-list 101 permit ip host X.X.X.X any
!
call rsvp-sync
!
voice-port 1/0/0
!
voice-port 1/0/1
!
voice-port 1/0/2
!
voice-port 1/0/3
!
voice-port 1/0/4
!
voice-port 1/0/5
!
voice-port 1/0/6
!
voice-port 1/0/7
!
mgcp
mgcp call-agent XXXX 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 999100 pots
application mgcpapp
destination-pattern 2599
port 1/0/0
!
!
!
call-manager-fallback
limit-dn 7910 1
limit-dn 7940 2
ip source-address X.X.X.X port 2000
max-ephones 48
max-dn 96
!
!
line con 0
logging synchronous
line aux 0
line vty 0 4
password cisco
login
!
!
end
09-24-2002 12:41 PM
I haven't set this up before, but I imagine you need a dial-peer defined for the fxs to make calls in h323 mode. I usually have a dial-peer with the destination-pattern of 9 pointing to the voice port that is connect to the PSTN.
09-24-2002 03:33 PM
Actually, with the 12.2.11 code, you need to add the following under the cal-manager-fallback to enable pstn access:
Call-manager-fallback
access-code pri 9 direct-inward-dial
access-code fxo 8
...
These are required for call-manager-fallback starting in 12.2.4 code.
09-25-2002 04:58 AM
Port 1/0/0 is an FXS port, not a trunk port. I don't think you need an access code for an FXS port. I may need them when I configure the PRI and T1 voice ports to the PSTN. I have a T1 for LD and a PRI for local/DID.
09-25-2002 06:11 AM
Have you tried adding a simple pots dial-peer with a destination of 9T pointing out the PRI. The fallback code supports the IP Phones but not the directly connected FXS port
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