ā11-08-2022 01:10 AM
Two Cisco Server's with CUCM11.5 & ISR 4321( E1 Modules)
for configuring 30 PRI/E1 which Trunk type has to be used SIP TRUNK / H.225
IF I Select SIP TRUNK with Route patterns incoming calls not working for outgoings no problem.
Solved! Go to Solution.
ā11-08-2022 06:33 AM - edited ā11-08-2022 08:41 AM
Hi,
From your first post it sounds like you have a SIP trunk setup in CUCM targetting the Gateway, is that correct?
If so, you have a few configuration additions to make, as shown below. That is also in addition to the advice Nithin provided above.
I saw that you have a voice translation rule configured, but not applied anywhere. Not a big deal if the caller ID delivered from your carrier matches your extension range, but if it doesn't then you need a translation applied somewhere (voice-port, dial-peer or do it in CUCM).
- Create an inbound dial-peer which allows calls to match this on the via header. e.g:
voice-class uri CUCM SIP
host ipv4:1.2.3.4
host ipv4:5.6.7.8
- Create a server-group to allow calls from the PSTN to match an outbound dial-peer to CUCM:
voice-class server-group 100
description ***CUCM Cluster***
ipv4: 1.2.3.4
ipv4: 5.6.7.8 preference 1
- Add an incoming dial-peer to match on the via header of the CUCM IP address:
dial-peer voice 100 voip
description Incoming to CUBE from CUCM
session protocol sipv2
incoming uri via CUCM
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
no vad
- Add an outbound dial-peer facing CUCM on your gateway:
dial-peer voice 200 voip
description Outgoing from CUBE to CUCM
session protocol sipv2
session server-group 100
voice-class codec 1
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
no vad
Also configure "forward-digits default" under dial-peer 1000. The "forward-digits 0" config is not needed there.
ā11-08-2022 10:55 PM
Here is a sample config Where incoming are through PRI and SIP trunk between CUCM and gateway
isdn switch-type primary-net5
!
!
trunk group VG-PRI-TRK
max-calls voice 60
voice service voip
ip address trusted list
ipv4 XX.XX.XX.XX
ipv4 XX.XX.XX.XX
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
trace
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
!
!
voice class uri 1 sip
host ipv4:XX.XX.XX.XX
host ipv4:XX.XX.XX.XX
!
voice class uri 2 sip
host ipv4:XX.XX.XX.XX
host ipv4:XX.XX.XX.XX
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
voice class server-group 1
ipv4 XX.XX.XX.XXpreference 1
ipv4 XX.XX.XX.XX preference 2
ipv4 XX.XX.XX.XX preference 1
ipv4 XX.XX.XX.XX preference 2
!
voice class sip-options-keepalive 1
!
!
!
!
voice translation-rule 1
rule 1 /^22087\(...$\)/ /7\1/
!
voice translation-rule 2
rule 1 /^\(7...\)$/ /2208\1/
!
!
voice translation-profile IN
translate called 1
!
voice translation-profile OUT
translate calling 2
!
!
controller E1 0/1/0
clock source line primary
pri-group timeslots 1-31
trunk-group VG-PRI-TRK timeslots 1-31
!
controller E1 0/1/1
clock source line secondary
pri-group timeslots 1-31
trunk-group VG-PRI-TRK timeslots 1-31
!
!
voice-port 0/1/0:15
translation-profile incoming IN
translation-profile outgoing OUT
!
voice-port 0/1/1:15
translation-profile incoming IN
translation-profile outgoing OUT
!
dial-peer voice 1 pots
description incoming calls from PSTN
incoming called-number .
direct-inward-dial
!
dial-peer voice 10 voip
description Outgoing calls to CUCM
destination-pattern ^74..$
session protocol sipv2
session server-group 1
voice-class codec 1
dtmf-relay rtp-nte sip-notify
!
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
!
dial-peer voice 23 pots
trunkgroup VG-PRI-TRK
description Service calls to PSTN
destination-pattern 91...
forward-digits 4
!
dial-peer voice 24 pots
trunkgroup VG-PRI-TRK
description Civil Defense calls to PSTN
destination-pattern 99999
forward-digits 4
!
ā11-08-2022 01:34 AM
for configuring 30 PRI/E1 which Trunk type has to be used SIP TRUNK / H.225:- its recommended to use SIP trunk as Ciscois removing support of H323 from latest ios versions.
IF I Select SIP TRUNK with Route patterns, that is for outgoing calls and not for incoming.
For incoming you must check the below things.
Dial-peers on the gateway
Voice translation from DID to your extensions.
Inbound CSS on trunk
if you share the voice gateway configurations we might be able to help you quickly.
ā11-08-2022 02:57 AM
ā11-08-2022 06:33 AM - edited ā11-08-2022 08:41 AM
Hi,
From your first post it sounds like you have a SIP trunk setup in CUCM targetting the Gateway, is that correct?
If so, you have a few configuration additions to make, as shown below. That is also in addition to the advice Nithin provided above.
I saw that you have a voice translation rule configured, but not applied anywhere. Not a big deal if the caller ID delivered from your carrier matches your extension range, but if it doesn't then you need a translation applied somewhere (voice-port, dial-peer or do it in CUCM).
- Create an inbound dial-peer which allows calls to match this on the via header. e.g:
voice-class uri CUCM SIP
host ipv4:1.2.3.4
host ipv4:5.6.7.8
- Create a server-group to allow calls from the PSTN to match an outbound dial-peer to CUCM:
voice-class server-group 100
description ***CUCM Cluster***
ipv4: 1.2.3.4
ipv4: 5.6.7.8 preference 1
- Add an incoming dial-peer to match on the via header of the CUCM IP address:
dial-peer voice 100 voip
description Incoming to CUBE from CUCM
session protocol sipv2
incoming uri via CUCM
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
no vad
- Add an outbound dial-peer facing CUCM on your gateway:
dial-peer voice 200 voip
description Outgoing from CUBE to CUCM
session protocol sipv2
session server-group 100
voice-class codec 1
voice-class sip options-keepalive
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
no vad
Also configure "forward-digits default" under dial-peer 1000. The "forward-digits 0" config is not needed there.
ā11-08-2022 10:11 PM - edited ā11-08-2022 10:13 PM
with provider we have 30 Channel PRI connected to E1 module of ISR 4321 Router.
ā11-08-2022 10:55 PM
Here is a sample config Where incoming are through PRI and SIP trunk between CUCM and gateway
isdn switch-type primary-net5
!
!
trunk group VG-PRI-TRK
max-calls voice 60
voice service voip
ip address trusted list
ipv4 XX.XX.XX.XX
ipv4 XX.XX.XX.XX
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
trace
sip
bind control source-interface GigabitEthernet0/0/0
bind media source-interface GigabitEthernet0/0/0
!
!
voice class uri 1 sip
host ipv4:XX.XX.XX.XX
host ipv4:XX.XX.XX.XX
!
voice class uri 2 sip
host ipv4:XX.XX.XX.XX
host ipv4:XX.XX.XX.XX
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
!
voice class server-group 1
ipv4 XX.XX.XX.XXpreference 1
ipv4 XX.XX.XX.XX preference 2
ipv4 XX.XX.XX.XX preference 1
ipv4 XX.XX.XX.XX preference 2
!
voice class sip-options-keepalive 1
!
!
!
!
voice translation-rule 1
rule 1 /^22087\(...$\)/ /7\1/
!
voice translation-rule 2
rule 1 /^\(7...\)$/ /2208\1/
!
!
voice translation-profile IN
translate called 1
!
voice translation-profile OUT
translate calling 2
!
!
controller E1 0/1/0
clock source line primary
pri-group timeslots 1-31
trunk-group VG-PRI-TRK timeslots 1-31
!
controller E1 0/1/1
clock source line secondary
pri-group timeslots 1-31
trunk-group VG-PRI-TRK timeslots 1-31
!
!
voice-port 0/1/0:15
translation-profile incoming IN
translation-profile outgoing OUT
!
voice-port 0/1/1:15
translation-profile incoming IN
translation-profile outgoing OUT
!
dial-peer voice 1 pots
description incoming calls from PSTN
incoming called-number .
direct-inward-dial
!
dial-peer voice 10 voip
description Outgoing calls to CUCM
destination-pattern ^74..$
session protocol sipv2
session server-group 1
voice-class codec 1
dtmf-relay rtp-nte sip-notify
!
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
!
dial-peer voice 23 pots
trunkgroup VG-PRI-TRK
description Service calls to PSTN
destination-pattern 91...
forward-digits 4
!
dial-peer voice 24 pots
trunkgroup VG-PRI-TRK
description Civil Defense calls to PSTN
destination-pattern 99999
forward-digits 4
!
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