[cisco-voip] SRST Prefix
Andrew Dignan
andy at dignans.com
Mon Feb 7 00:33:23 EST 2005
The key to the "Alias" command working properly is to configure "no
huntstop" under the telephony service. Below is a working config of
SRST/MGCP Fall back that works with no issues. Notice the incoming DNIS
is 4205, which I change to 6530. The VOIP dial peer is 4205, which I
change to 6530, which in turn is the default destination/first ALIAS call.
Lastly, ensure you have the "call application alternate DEFAULT" command
in or else you will hear dead air.
*******************************************
voice rtp send-recv
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g729br8
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
ccm-manager fallback-mgcp
ccm-manager redundant-host x.x.x.x
ccm-manager mgcp
ccm-manager music-on-hold
mta receive maximum-recipients 0
!
!
controller T1 0/0
framing esf
linecode b8zs
ds0-group 1 timeslots 1-24 type e&m-wink-start
description VWIC-1MFT-T1 PSTN CAS T1
!
class-map match-all VoIP
match access-group name VoIP-ACL
class-map match-all VoIP-Control
match access-group name VoIP-Control-ACL
class-map match-all VoIP-CTIos
match access-group name VoIP-CTIos-ACL
!
!
policy-map VoIP-CTIos-LLQ
class VoIP
priority 320
class VoIP-Control
bandwidth 24
class VoIP-CTIos
bandwidth 32
class class-default
fair-queue
!
translation-rule 1
Rule 1 ^4205 6530
!
!
!
!
interface Loopback0
ip address x.x.x.x 255.255.255.255
ip pim sparse-mode
!
interface FastEthernet0/0
description *****Connection to ROUTER*****
no ip address
duplex full
speed 100
!
interface FastEthernet0/0.5
description Data Vlan 5
encapsulation dot1Q 5 native
ip address x.x.x.x 255.255.255.0
ip helper-address x.x.x.x
!
interface FastEthernet0/0.105
description Voice Vlan 105
encapsulation dot1Q 105
ip address x.x.x.x 255.255.255.0
ip helper-address x.x.x.x
ip pim sparse-mode
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/1
description *****PTP to xxx*****
ip address x.x.x.x 255.255.255.252
ip pim sparse-mode
service-policy output VoIP-CTIos-LLQ
!
router eigrp 100
passive-interface Loopback0
network x.x.x.x
no auto-summary
no eigrp log-neighbor-changes
!
no ip http server
ip classless
ip route 0.0.0.0 0.0.0.0 x.x.x.x
!
ip pim rp-address x.x.x.x
!
!
ip access-list extended TestSRST
deny tcp any x.x.x.x 0.0.0.255 eq 2000
deny udp any x.x.x.x 0.0.0.255 eq 2000
deny udp any x.x.x.x 0.0.0.255 eq 2427
deny tcp any x.x.x.x 0.0.0.255 eq 2427
permit ip any any
ip access-list extended VoIP-ACL
permit ip any any dscp ef
ip access-list extended VoIP-Control-ACL
permit ip any any dscp af31
!
!
!
call application alternate DEFAULT
call rsvp-sync
!
voice-port 0/0:1
input gain -3
output attenuation 3
echo-cancel coverage 32
description *****CID# xxxxx*****
!
voice-port 1/1/0
echo-cancel coverage 32
echo-cancel suppressor
no comfort-noise
timeouts wait-release 3
timing hookflash-out 50
description *****POTS BACKUP*****
music-threshold -70
!
voice-port 1/1/1
echo-cancel coverage 32
echo-cancel suppressor
no comfort-noise
timeouts wait-release 3
timing hookflash-out 50
description *****OVERHEAD PAGING*****
music-threshold -70
!
mgcp
mgcp call-agent x.x.x.x 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
no mgcp package-capability res-package
mgcp package-capability sst-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp bind control source-interface Loopback0
mgcp bind media source-interface Loopback0
mgcp rtp payload-type g726r16 static
!
mgcp profile default
!
!
!
dial-peer cor custom
!
!
!
dial-peer voice 999110 pots
application mgcpapp
port 1/1/0
!
dial-peer voice 999111 pots
application mgcpapp
port 1/1/1
!
dial-peer voice 999001 pots
application mgcpapp
port 0/0:1
!
dial-peer voice 9 pots
description *****7-Digit - SRST*****
application default
destination-pattern 9[2-9]......
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0:1
forward-digits 7
!
dial-peer voice 91 pots
description *****10-Digit - SRST*****
application default
destination-pattern 91[2-9].........
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 0/0:1
!
dial-peer voice 911 pots
description *****911 - SRST*****
application default
destination-pattern 9911
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
port 1/1/0
forward-digits 3
!
dial-peer voice 80 pots
description *****Overhead Paging*****
destination-pattern #80
port 1/1/1
!
dial-peer voice 4205 voip
tone ringback alert-no-PI
preference 1
application default
destination-pattern 4205
progress_ind setup enable 3
translate-outgoing called 1
voice-class codec 1
session target ipv4:x.x.x.x
no vad
!
!
call-manager-fallback
max-conferences 4
limit-dn 7910 1
limit-dn 7940 2
limit-dn 7960 6
timeouts interdigit 3
ip source-address x.x.x.x port 2000
max-ephones 48
max-dn 192
keepalive 10
default-destination 6530
voicemail xxxx
no huntstop
alias 1 6530 to 6530 preference 1
alias 2 6510 to 6536 preference 2
date-format dd-mm-yy
!
!
line con 0
password 7
stopbits 1
line aux 0
line vty 0 4
password 7
login
line vty 5 15
password 7
login
!
ntp clock-period 17180208
ntp server x.x.x.x prefer
!
end
*********************************************
Andy - Berbee
> Gents,
>
> Thanks for the prompt replies. The 'attendant DN' is configured properly
> under the port configuration in CCMAdmin. Inbound/outbound calls work
> fine
> under normal operations; inbound calls in fallback mode are a problem
> though.
>
> The extension that is configured under the 'attendant DN' is the pilot
> number for my Unity 'Auto Attendant' call handler application. However,
> in
> fallback mode (SRST) I cannot redirect inbound calls to an alternate
> extension because the 'attendant DN' (auto attendant) is no longer
> available?
>
> I have tried using the 'Alias' command. When I enable my call trace
> (debug
> voip ccapi inout) I see the call come into the gateway and connect, but
> then
> nothing happens (dead air). I even tried using translation profiles to
> convert the 'called number', or lack thereof, to a known extension -
> similar
> behaviour to 'Connection Plar' for H.323. Same 'dead air' results.
>
> Please keep the feedback coming, they're all great ideas.
>
> Best regards,
> Colin
>
>
>
> -----Original Message-----
> From: Wes Sisk [mailto:wsisk at cisco.com]
> Sent: Sunday, February 06, 2005 5:43 PM
> To: Colin Lowe
> Cc: 'Ryan Ratliff'; cisco-voip at puck.nether.net; 'King, Jesse'
> Subject: Re: [cisco-voip] SRST Prefix
>
> For MGCP, you will need to configure "attendant dn" under the port
> configuration in CCMAdmin.
>
> Step3:
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
> a008009428e.shtml
>
> /Wes
>
> Colin Lowe wrote:
>
>>Ryan,
>>
>>I can see how this will work where the gateway is a non-MGCP controlled
>>gateway (i.e. H.323).
>>
>>However, if the gateway is an MGCP gateway, how do you redirect the call
>> in
>>SRST mode, while not affecting inbound calls during normal operation? If
> I'm
>>reading my traces properly, application MGCPAPP seems to invoke and the
> call
>>goes nowhere (I hear dead air).
>>
>>Any help would be greatly appreciated...
>>
>>Colin
>>
>>
>>
>>-----Original Message-----
>>From: cisco-voip-bounces at puck.nether.net
>>[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of King, Jesse
>>Sent: Friday, February 04, 2005 3:30 PM
>>To: Ryan Ratliff
>>Cc: cisco-voip at puck.nether.net
>>Subject: RE: [cisco-voip] SRST Prefix
>>
>>Exactly what I was looking for. Worked perfect. Thanks.
>>
>>Jesse
>>
>>-----Original Message-----
>>From: Ryan Ratliff [mailto:rratliff at cisco.com]
>>Sent: Friday, February 04, 2005 3:03 PM
>>To: King, Jesse
>>Cc: cisco-voip at puck.nether.net
>>Subject: Re: [cisco-voip] SRST Prefix
>>
>>For an FXO port you'll have to configure a connection plar <dn> under the
>>voice-port for the FXO port.
>>
>>Since there is no concept of a called number with FXO you have to
>> redirect
>>every call inbound on the port to a specific place.
>>
>>-Ryan
>>On Feb 4, 2005, at 2:36 PM, King, Jesse wrote:
>>
>>Can someone point me in the right direction to point inbound calls with
> SRST
>>on an FXO port to a specific extension?
>>
>>Thanks.
>>
>>Jesse
>>_______________________________________________
>>cisco-voip mailing list
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>>
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>>
>>
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