ā01-11-2023 07:05 AM
Hi,
I'm quite new to the Cisco unified communications manager. In my premises I have two PRI connection at gateway
Primary PRI => 022 4303 "extension number" for Incoming calls and we need dial 90 "mobile number" for outgoing calls with reference to my Cisco up phone's
In similar way for
Secondary PRI => we need dial from Cisco ip *90 "mobile number" for outgoing calls.
But I want to configure my gateway in such way so that it can automatically switch between PRI lines if any one PRI is down, so users doesn't need to change the dailing pattern if any one PRI is down or not working
Requesting you to please suggest how I'll able to achieve this, I'm sure there are some parameters which I need configure in gateway but I don't know how to do it, please any one guide me
Solved! Go to Solution.
ā01-11-2023 12:37 PM
if you can share the gateway configurations we can provide you exact configurations. I would prefer to go with the TrunkGroup. Here mentioned is a sample configurations. And i would strip the extra number from CUCM to make dial-peers simple.
trunk group VG-PRI-TRK
controller E1 0/1/0
framing no-crc4
clock source line secondary
pri-group timeslots 1-31
trunk-group VG-PRI-TRK timeslots 1-31
!
controller E1 0/1/1
framing no-crc4
clock source line primary
pri-group timeslots 1-31
trunk-group VG-PRI-TRK timeslots 1-31
dial-peer voice 20 pots
trunkgroup VG-PRI-TRK
description Local calls to PSTN
destination-pattern 92.......
forward-digits 8
!
dial-peer voice 21 pots
trunkgroup VG-PRI-TRK
description Mobile calls to PSTN
destination-pattern 9[789].......
forward-digits 8
!
dial-peer voice 22 pots
trunkgroup VG-PRI-TRK
description International calls to PSTN
destination-pattern 900T
prefix 00
Note:- instead of channel assigning to the Trunk group you can do the D channel using below command.
Interface S0/0/1:15
trunkgroup VG-PRI-TRK
ā06-29-2023 04:36 AM
As youāre using MGCP to control your ISDN circuits youāll do what you ask for in CM. The dial peer configuration is not applicable to this.
ā07-03-2023 02:11 AM - edited ā07-04-2023 12:13 AM
Reading what you wrote once more I see that you're using two different route patterns, 90 that sends calls to the gateway and uses the SIP trunk to you service provider and *90 that uses the MGCP controlled PRIs. To get an automatic switch between these two you would configure this.
That's all there is to this, assuming that your gateway configuration is correct.
On the dial peers I would based on the configuration you shared recommend that you do these changes.
no dial-peer voice 102 voip !not needed as dial peer 101 is identical
no dial-peer voice 202 voip !not needed as dial peer 201 is identical
!
dial-peer voice 101 voip
destination-pattern .T !Change this to something more specific than .T as that matches anything. I would put it as 90T if your using that as the common route string
!
dial-peer voice 201 voip
no destination-pattern +91.T !not needed on an inbound dial peer
no session target ipv4:xx.xx.xx.xx !not needed on an inbound dial peer
no session transport tcp !as the outbound dial peer does not have TCP set it is not very likely that the inbound would need this
!
dial-peer voice 400 voip
no session transport tcp
no incoming called-number .T
!Apart from the changes to the dial peers you also need to turn on the Cube functionallity in the gateway for it to act as an SBC. Do that with this and also turn back on the security feature in the gateway.
voice service voip
ip address trusted list
ipv4 ITSP IP 1
ipv4 ITSP IP 2
ipv4 CM IP 1
ipv4 CM IP 2
address-hiding
mode border-element license periodicity days 30
ā07-30-2023 07:38 AM - edited ā07-30-2023 12:12 PM
You have not provided the RL/RG details. Please open the route list details on each of the route groups that are included in the route lists and add that information to your document.
What is clear so far is this, on the route pattern used for sending calls to the PRI, ie *9.! you have a discard digit set to PreDot for the called number and your prefixing +226925 to the calling number. So this is what you need to put on the RL/RG level in the route list where you have the RG which holds your MGCP PRI as the second option so that your sending called and calling numbers in the format as your service provider expects it. Also on the other route pattern, the one used primarily for the SIP trunk, you are prefixing the calling number with +91224303. This as well needs to be put on the RL/RG level on the RG used for the SIP trunk as you cannot use the settings on the RP level if you want to achieve what you ask about.
On the MGCP PRI youāre using a CSS for called and calling number transformation named css-ISD. You need to look at what partition(s) that you have in that CSS and what transformation(s) that you have in these partition(s) as that would as well influence how you send these numbers to your service provider.
ā07-31-2023 11:18 PM
Glad to hear that! However what youāre describing doesnāt make any sense at all as if the problem was caused by missing partition(s) in a calling search space calls via the route pattern that goes via the SIP trunk, ie the first option in the route list, would not have worked either.
Another thing, although not directly related to the problem youāve been working on, by what you shared in the documents you have a CSS set for calling/called/redirecting number transformation. As that CSS, named ācss-ISDā, doesnāt include any partition(s) used for transformation there is no need to have that set for transformation CSSās on the MGCP PRIs. Recommend you to clear those settings and reset the MGCP trunks.
Add on bonus information, you should never ever mix call routing with transformations as that will lead to undesirable results. Any partition(s) and CSS(s) used for transformation should not include any selected partition(s) that holds configuration elements used for call routing, such as translation pattern(s), route pattern(s) or directory number(s).
ā01-11-2023 07:20 AM
Just use the same dial-peer twice, with different PRI ports and different preferences:
dial-peer 1 pots
preference 1
destination-pattern 90
port 0/0/0:15
!
dial-peer 2 pots
preference 2
destination-pattern 90
port 0/0/1:15
ā01-11-2023 10:28 AM
As the OP is using different call patterns for the two PRIs that would not work, at least not without additional configuration.
PRI 1: 90 + "mobile number"
PRI 2: *90 + "mobile number"
So second PRI has a star as the start.
ā01-11-2023 10:37 AM
Yeah, that's what he currently has.
But he wrote, that he doesn't want to have that anymore and use both PRI's with only one pattern 90:
"But I want to configure my gateway in such way so that it can automatically switch between PRI lines if any one PRI is down, so users doesn't need to change the dailing pattern if any one PRI is down or not working"
ā01-11-2023 11:04 AM
There are different ways that this part can be understood. I read it as the OP wants to keep the current dial patterns, but have the calls be able to use either of the PRIs by automatically adjusting the called number.
ā06-29-2023 04:24 AM
Thank you so much sir for your support, I have shared my gateway configuration requesting you to please share the what are the updation required in current configurations
incoming uri via cucm
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
!
dial-peer voice 400 voip
call-block translation-profile incoming Block-incoming
call-block disconnect-cause incoming call-reject
session protocol sipv2
session transport tcp
incoming called-number .T
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
!
!
!
gatekeeper
shutdown
!
!
GW01# exit
ā01-11-2023 07:21 AM
what do you have for dial-peers?
ā01-11-2023 10:35 AM
There a multiple ways to achieve what you ask about. One way would be to use called party transformation on the RL/RG level to add the star to the path where the second PRI is present for when the caller called 90 "mobile number". Then on the RL/RG level used for the *90 RP do a discard digit on the path to the first PRI for when the caller called *90 "mobile number" to drop the star.
ā01-11-2023 12:37 PM
if you can share the gateway configurations we can provide you exact configurations. I would prefer to go with the TrunkGroup. Here mentioned is a sample configurations. And i would strip the extra number from CUCM to make dial-peers simple.
trunk group VG-PRI-TRK
controller E1 0/1/0
framing no-crc4
clock source line secondary
pri-group timeslots 1-31
trunk-group VG-PRI-TRK timeslots 1-31
!
controller E1 0/1/1
framing no-crc4
clock source line primary
pri-group timeslots 1-31
trunk-group VG-PRI-TRK timeslots 1-31
dial-peer voice 20 pots
trunkgroup VG-PRI-TRK
description Local calls to PSTN
destination-pattern 92.......
forward-digits 8
!
dial-peer voice 21 pots
trunkgroup VG-PRI-TRK
description Mobile calls to PSTN
destination-pattern 9[789].......
forward-digits 8
!
dial-peer voice 22 pots
trunkgroup VG-PRI-TRK
description International calls to PSTN
destination-pattern 900T
prefix 00
Note:- instead of channel assigning to the Trunk group you can do the D channel using below command.
Interface S0/0/1:15
trunkgroup VG-PRI-TRK
ā06-28-2023 05:32 AM
Dear Sir,
Thank you for your help
Here are the my gateway configuration details, still need to set priority so the calls can be auto routed through the active PRI link
no service password-encryption
!
hostname GW01
!
boot-start-marker
boot-end-marker
!
aqm-register-fnf
!
card type e1 0 0
card type e1 0 1
card type e1 0 2
card type e1 0 3
!
no aaa new-model
clock timezone IST 5 30
network-clock-participate wic 0
network-clock-participate wic 1
network-clock-participate wic 2
network-clock-participate wic 3
!
!
!
!
no ip domain lookup
ip cef
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-4ess
!
!
voice-card 0
dspfarm
dsp services dspfarm
!
!
!
voice service voip
no ip address trusted authenticate
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
asserted-id ppi
early-offer forced
!
!
voice class uri cucm sip
host ipv4:xx.xx.xx.xx
host ipv4:xx.xx.xx.xx
host ipv4:xx.xx.xx.xx
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729br8
!
voice class sip-profiles 1
request INVITE sip-header SIP-Req-URI modify "@xx.xx.xx.xx" "@abc.xyz.in"
request INVITE sip-header From modify "<>" "<>@abc.xyz.in>"
request INVITE sip-header P-Preferred-Identity modify "@(.*)" "abc.xyz.in>"
!
!
voice iec syslog
!
!
voice translation-rule 1
rule 3 /^9/ //
!
voice translation-rule 2
rule 1 /^912243031/ /1/
rule 2 /^912243032/ /2/
rule 3 /^912243033/ /3/
rule 4 /^912243034/ /4/
rule 5 /^912243035/ /5/
rule 6 /^912243036/ /6/
rule 7 /^912243037/ /7/
rule 8 /^912243038/ /8/
!
voice translation-rule 3
rule 1 /^0\(.*\)/ /90\1/
!
voice translation-rule 20
rule 1 reject /09636590486/
rule 2 reject /07727006044/
rule 3 reject /08956339827/
rule 4 reject /08961434009/
!
!
voice translation-profile Block-incoming
translate calling 20
!
voice translation-profile NineRemove
translate called 1
!
voice translation-profile incoming_call4digit
translate called 2
!
voice translation-profile zero
translate calling 3
!
!
!
license udi pid CISCO2951/K9 sn FGL190910RP
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!
hw-module pvdm 0/2
!
!
!
!
redundancy
!
!
controller E1 0/0/0
pri-group timeslots 1-31 service mgcp
description TATA PRI
!
controller E1 0/0/1
pri-group timeslots 1-31 service mgcp
!
controller E1 0/1/0
pri-group timeslots 1-31 service mgcp
!
controller E1 0/1/1
pri-group timeslots 1-31 service mgcp
!
controller E1 0/1/2
pri-group timeslots 1-31 service mgcp
!
controller E1 0/1/3
pri-group timeslots 1-31 service mgcp
!
controller E1 0/2/0
pri-group timeslots 1-31 service mgcp
!
controller E1 0/2/1
pri-group timeslots 1-31 service mgcp
!
controller E1 0/2/2
pri-group timeslots 1-31 service mgcp
!
controller E1 0/2/3
pri-group timeslots 1-31 service mgcp
!
controller E1 0/3/0
pri-group timeslots 1-31 service mgcp
!
controller E1 0/3/1
pri-group timeslots 1-31 service mgcp
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
ip address xx.xx.xx.xx xx.xx.xx.xx
duplex auto
speed auto
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
interface Serial0/0/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
interface Serial0/1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
interface Serial0/1/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
interface Serial0/1/2:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
interface Serial0/1/3:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
interface Serial0/2/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
interface Serial0/2/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
interface Serial0/2/2:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
interface Serial0/2/3:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
interface Serial0/3/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn timer T310 120000
isdn not-end-to-end 64
isdn protocol-emulate network
isdn incoming-voice voice
isdn map address .* plan isdn type national
isdn bind-l3 ccm-manager
isdn send-alerting
no cdp enable
!
interface Serial0/3/1:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
ip forward-protocol nd
!
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
ip route 0.0.0.0 0.0.0.0 xx.xx.xx.xx
!
!
nls resp-timeout 1
cpd cr-id 1
cpd
!
!
control-plane
!
!
voice-port 0/0/0:15
cptone IN
!
voice-port 0/1/0:15
cptone IN
!
voice-port 0/2/0:15
!
voice-port 0/3/0:15
cptone IN
bearer-cap Speech
!
voice-port 0/0/1:15
cptone IN
!
voice-port 0/1/1:15
!
voice-port 0/2/1:15
!
voice-port 0/3/1:15
!
voice-port 0/1/2:15
!
voice-port 0/2/2:15
!
voice-port 0/1/3:15
!
voice-port 0/2/3:15
!
!
!
!
!
ccm-manager redundant-host xx.xx.xx.xx xx.xx.xx.xx
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server xx.xx.xx.xx xx.xx.xx.xx
ccm-manager config
!
mgcp
mgcp call-agent xx.xx.xx.xx 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp bind control source-interface GigabitEthernet0/0
mgcp bind media source-interface GigabitEthernet0/0
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm xx.xx.xx.xx identifier 3 priority 3 version 7.0
sccp ccm xx.xx.xx.xx identifier 1 priority 1 version 7.0
sccp ccm xx.xx.xx.xx identifier 2 priority 2 version 7.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register Transcode
!
sccp ccm group 2
bind interface GigabitEthernet0/0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register Conference
!
dspfarm profile 1 transcode
codec g729br8
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 2
associate application SCCP
!
dspfarm profile 2 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 1
associate application SCCP
!
dial-peer voice 101 voip
description ** outgoing **
translation-profile outgoing NineRemove
destination-pattern .T
session protocol sipv2
session target ipv4:xx.xx.xx.xx
voice-class codec 1 offer-all
voice-class sip asserted-id ppi
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte sip-notify sip-kpml
!
dial-peer voice 201 voip
description ** incoming **
translation-profile outgoing zero
destination-pattern +91.T
session protocol sipv2
session target ipv4:xx.xx.xx.xx
session transport tcp
incoming uri via cucm
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte sip-notify sip-kpml
!
dial-peer voice 102 voip
description ** outgoing **
translation-profile outgoing NineRemove
destination-pattern .T
session protocol sipv2
session target ipv4:xx.xx.xx.xx
voice-class codec 1 offer-all
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte sip-notify sip-kpml
!
dial-peer voice 202 voip
description ** incoming **
translation-profile outgoing zero
destination-pattern +91.T
session protocol sipv2
session target ipv4:xx.xx.xx.xx
incoming uri via cucm
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
!
dial-peer voice 400 voip
call-block translation-profile incoming Block-incoming
call-block disconnect-cause incoming call-reject
session protocol sipv2
session transport tcp
incoming called-number .T
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
!
!
!
gatekeeper
shutdown
!
!
GW01# exit
ā06-29-2023 04:36 AM
As youāre using MGCP to control your ISDN circuits youāll do what you ask for in CM. The dial peer configuration is not applicable to this.
ā06-29-2023 05:22 AM
ā06-29-2023 06:36 AM - edited ā06-29-2023 08:14 AM
Not rocket science really. You just configure route pattern (RP) and point it to a route list (RL), that contains one or multiple route groups (RG) depending on your specific needs. The RG contains one or many MGCP controlled circuits. If you need to do modifications to either called or calling numbers based on the different circuits used you could do that on the RL/RG level, so then you'd use multiple RG that you put into a RL.
ā06-29-2023 07:01 AM
ā07-02-2023 09:59 PM
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