02-18-2014 11:51 PM - edited 03-18-2019 11:20 AM
Dear All,
I have a scenario in Lab where I have configured two clusters. Please see the below scenario.
All the phones in 1XX are registered to CUCM and 2XXX are registered to CME and can call to each other in the same cluster.
Now I want to make calls from 1XXX to 2XXX, so what will be dial peer I need to configure in the routers.
Regards
Amarjit Das
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02-19-2014 03:18 AM
Hi Amarjit.
You are missing some details on your configs.
On GW
voice service voip
allow connections sip-to-sip
dial-peer 11
+++you have+++
session target ipv4:10.113.113.2
+++ based on your diagram CUCM IP address should be 10.0.0.1 and session target should point to it+++
session target ipv4:10.0.0.1
In your voice class codec you have selected g729 codec only.
Which codec have you defined on Region configuration on CUCM?
This should match the same codec
Pease check the region associated to device pool defined for the GW.
to simplify this operation allow codec g711ulaw on both CME and GW because is the default codec negotiated by default region on CUCM.
To achieve this, modify voice class codec on both GW and CME:
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729br8
++++ON CME+++
Remove dialpeer 11( you don't need it because extensions are locally registered)
HTH
Regards
Carlo
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02-19-2014 06:12 AM
Hi Das,
can u try adding session protocol sipv2 in CME router under
dial-peer voice 10 voip
destination-pattern 1...
voice-class codec 1
session target ipv4:172.16.0.1
regds,
aman
02-19-2014 12:11 AM
Hello Amarjit,
In CUCM, you can configure a SIP Trunk with CME as destination IP. configure RP with 2XXX and assign the SIP trunk to it.
in CME, configure voip dial-peer to receive the call from CUCM.
and configure another dial-peer with dest-pattern: 1... and session target
Ensure you have proper IP routing between these devices and confirm the reachability.
config example below
!
voice service voip
sip
bind control source-interface <.2 interface>
bind media source-interface <.2 interface>
!
dial-peer voice 10 voip
description **SIP TRUNK from CUCM**
incoming called-number 2...
session protocol sipv2
dtmf-relay rtp-nte
no vad
!
!
dial-peer voice 11 voip
description **SIP TRUNK to CUCM**
destination-pattern 1...
session protocol sipv2
session target ipv4:10.0.0.1
dtmf-relay rtp-nte
no vad
!
//Suresh
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02-19-2014 12:20 AM
Hi Amarjit.
First I would suggest you to change signaling protocol between CUCM cluster and the Voice gateway from H323 to Sip.
In your scenario coul be enough a route pattern in CUCM such as 2XXX pointing to the gateway.
Than on H323 Gateway:
voice class codec 1
codec preference 1 g711ulaw
codec preference 1 g711alaw
codec preference 1 g728br8
dial-peer voice 10 voip
description ++++TO CME++++
destination-pattern 2...
session target ipv4:192.168.0.2
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 11 voip
description ++++TO CUCM++++
destination-pattern 1...
session target ipv4:10.0.0.1
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 12 voip
description ++++Incoming From CME++++
incoming called-number 1...
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
In the same way on CME configure dialpeers pointing back to H323 Gateway.
HTH
Regards
Carlo
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"The more you help the more you learn"
02-19-2014 12:49 AM
Hi Carlo,
Thanks for your reply.
I will try the same you have said.
But can you tell me what is configuration required to configure the gateway as SIP.
Regards
Amarjit Das
02-19-2014 12:54 AM
Here is the config to enable to SIP on the GW:
voice service voip
sip
bind control source-interface < interface>
bind media source-interface
!
//Suresh
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02-19-2014 01:15 AM
Thank Suresh,
Let me try with the said configuration.
Vil let u know..
Regards
Amarjit Das
02-19-2014 01:17 AM
Hi Amarjit.
In addition to what Suresh mentioned (+5) you should configure a sip trunk on CUCM specifing the ip address of the gateway and modify dialpeers on both Gateway and CME specifyng sip as session protocol.
Eg.
dial-peer voice 10 voip
description ++++TO CME++++
destination-pattern 2...
session protocol sipv2
session target ipv4:192.168.0.2
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 11 voip
description ++++TO CUCM++++
session protoco sipv2
destination-pattern 1...
session target ipv4:10.0.0.1
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 12 voip
description ++++Incoming From CME++++
session protocol sipv2
incoming called-number 1...
voice-class codec 1
dtmf-relay rtp-nte
no vad
Remember also to allow sip to sip transactions by configuring
voice service voip
allow connections sip-to-sip
..and last from IOS version 15.1(2) toll fraud prevetion was introduced by default and asks you to specify ip address allowed to initiate a SIP or H323 session with your VG.
You can configure it through:
voice service voip
ip address trusted list
ipv4 x.x.x.x y.y.y.y where x.x.x.x is the subnet or the host you want to allow and y.y.y.y is the subnet mask.
To disable this feature:
voice service voip
no ip address trusted list
HTH
Regards
Carlo
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02-19-2014 01:26 AM
Hi Amarjit,
[+5] for Carlo for inputs.
can u share the config on CME& Gateway?
regds,
aman
02-19-2014 02:54 AM
Hi,
I have configured both the routers as said above but still call is not established.
Please find the configuration below ---
VOICE-GW
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname VOICE-GW
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
memory-size iomem 5
ip cef
!
ip auth-proxy max-nodata-conns 3
ip admission max-nodata-conns 3
!
multilink bundle-name authenticated
!
voice service voip
sip
session transport tcp
!
!
voice class codec 1
codec preference 1 g729br8
!
!
archive
log config
hidekeys
!
interface FastEthernet0/0
ip address 10.113.113.4 255.255.255.0
duplex auto
speed auto
!
interface Serial0/0
no ip address
shutdown
clock rate 2000000
!
interface FastEthernet0/1
ip address 172.16.0.1 255.255.255.252
duplex auto
speed auto
!
interface Serial0/1
no ip address
shutdown
clock rate 2000000
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 FastEthernet0/1
!
!
ip http server
no ip http secure-server
!
control-plane
!
dial-peer voice 10 voip
destination-pattern 2...
voice-class codec 1
session protocol sipv2
session target ipv4:172.16.0.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 11 voip
destination-pattern 1...
voice-class codec 1
session protocol sipv2
session target ipv4:10.113.113.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 12 voip
voice-class codec 1
session protocol sipv2
incoming called-number 1...
dtmf-relay rtp-nte
no vad
CME
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname CME
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
memory-size iomem 5
!
!
ip cef
no ip domain lookup
ip domain name lab.local
!
!
multilink bundle-name authenticated
!
voice class codec 1
codec preference 1 g729br8
!
archive
log config
hidekeys
!
interface FastEthernet0/0
ip address 172.16.0.2 255.255.255.252
duplex auto
speed auto
!
interface FastEthernet0/1
ip address 192.168.0.2 255.255.255.0
duplex auto
speed auto
!
!
no ip http server
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 FastEthernet0/0
!
control-plane
dial-peer voice 10 voip
destination-pattern 1...
voice-class codec 1
session target ipv4:172.16.0.1
!
dial-peer voice 11 voip
destination-pattern 2...
voice-class codec 1
session target ipv4:192.168.0.2
!
dial-peer voice 12 voip
voice-class codec 1
incoming called-number 2...
no vad
!
!
telephony-service
max-ephones 2
max-dn 2
ip source-address 192.168.0.2 port 2000
auto assign 1 to 2
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1
number 2001
!
!
ephone-dn 2
number 2002
!
!
ephone 1
no multicast-moh
mac-address 001F.CA4C.6E03
keepalive 30 auxiliary 0
type 7941
button 1:1
!
!
!
ephone 2
no multicast-moh
keepalive 30 auxiliary 0
Please suggest.
Regards
Amarjit Das
02-19-2014 03:18 AM
Hi Amarjit.
You are missing some details on your configs.
On GW
voice service voip
allow connections sip-to-sip
dial-peer 11
+++you have+++
session target ipv4:10.113.113.2
+++ based on your diagram CUCM IP address should be 10.0.0.1 and session target should point to it+++
session target ipv4:10.0.0.1
In your voice class codec you have selected g729 codec only.
Which codec have you defined on Region configuration on CUCM?
This should match the same codec
Pease check the region associated to device pool defined for the GW.
to simplify this operation allow codec g711ulaw on both CME and GW because is the default codec negotiated by default region on CUCM.
To achieve this, modify voice class codec on both GW and CME:
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729br8
++++ON CME+++
Remove dialpeer 11( you don't need it because extensions are locally registered)
HTH
Regards
Carlo
Please rate all helpful posts
"The more you help the more you learn"
02-19-2014 03:58 AM
Hi Carlo,
Thank for your reply,
Now I can call from 1XXX to 2XXX, but from 2XXX to 1XXX call is not happening.
Please suggest.
02-19-2014 04:05 AM
Hi Amarjit.
Did you check incoming CSS on configured sip trunk on CUCM?
Please send the actual config of both CME and Gateway.
Please activate a debug voip dialpeer inout on both, make a call and send the output.
Thanks
Regards
Carlo
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"The more you help the more you learn"
02-19-2014 04:25 AM
Hi Carlo,
I have kept default device pool and CSS.
Please find the config below --
Voice Gateway --
voice service voip
allow-connections sip to sip
sip
session transport tcp
!
!
voice class codec 1
codec preference 1 g711ulaw
!
interface FastEthernet0/0
ip address 10.113.113.4 255.255.255.0
duplex auto
speed auto
!
interface Serial0/0
no ip address
shutdown
clock rate 2000000
!
interface FastEthernet0/1
ip address 172.16.0.1 255.255.255.252
duplex auto
speed auto
!
interface Serial0/1
no ip address
shutdown
clock rate 2000000
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 FastEthernet0/1
!
dial-peer voice 10 voip
destination-pattern 2...
voice-class codec 1
session protocol sipv2
session target ipv4:172.16.0.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 11 voip
destination-pattern 1...
voice-class codec 1
session protocol sipv2
session target ipv4:10.113.113.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 12 voip
voice-class codec 1
session protocol sipv2
incoming called-number 1...
dtmf-relay rtp-nte
no vad
CME Router
voice class codec 1
codec preference 1 g711ulaw
!
!
interface FastEthernet0/0
ip address 172.16.0.2 255.255.255.252
duplex auto
speed auto
!
interface FastEthernet0/1
ip address 192.168.0.2 255.255.255.0
duplex auto
speed auto
!
!
no ip http server
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 FastEthernet0/0
!
dial-peer voice 10 voip
destination-pattern 1...
voice-class codec 1
session target ipv4:172.16.0.1
!
dial-peer voice 12 voip
voice-class codec 1
incoming called-number 2...
no vad
!
!
telephony-service
max-ephones 2
max-dn 2
ip source-address 192.168.0.2 port 2000
auto assign 1 to 2
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1
number 2001
!
!
ephone-dn 2
number 2002
!
!
ephone 1
no multicast-moh
mac-address 001F.CA4C.6E03
keepalive 30 auxiliary 0
type 7941
button 1:1
!
!
!
ephone 2
no multicast-moh
keepalive 30 auxiliary 0
!
Note : Please dont worry about the IP address as I have made some changes in the IP's
Now when I am calling from 1001 to 2001 call is establishing, but when I am calling from 2001 to 1001 call is not establishing.
So I have collected the logs after calling from 2001 to 1001 on both the routers.
Voice Gateway debug log --
*Mar 1 02:40:45.371: //-1/2B88C4988041/DPM/dpAssociateIncomingPeerCore:
Calling Number=2001, Called Number=1001, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 02:40:45.379: //-1/2B88C4988041/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=12
*Mar 1 02:40:45.395: //-1/2B88C4988041/DPM/dpAssociateIncomingPeerCore:
Calling Number=2001, Called Number=1001, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 02:40:45.399: //-1/2B88C4988041/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=12
*Mar 1 02:40:45.459: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=1001, Called Number=1001, Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 02:40:45.463: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=1001
*Mar 1 02:40:45.467: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Mar 1 02:40:45.471: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=11
*Mar 1 02:40:45.475: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=1001, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 02:40:45.475: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=11
*Mar 1 02:40:45.483: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Calling Number=, Called Number=1001, Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 02:40:45.483: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=1001
*Mar 1 02:40:45.483: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Mar 1 02:40:45.483: //-1/2B88C4988041/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=11
CME Router Debug Logs--
*Mar 1 01:47:49.155: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=2001, Called Number=, Voice-Interface=0x65ECD080,
Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 01:47:49.159: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20001
*Mar 1 01:47:49.675: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Calling Number=, Called Number=1, Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 01:47:49.679: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=1
CME#
*Mar 1 01:47:49.679: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Mar 1 01:47:49.683: //-1/2B88C4988041/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Mar 1 01:47:49.867: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Calling Number=, Called Number=10, Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 01:47:49.871: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=10
*Mar 1 01:47:49.871: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Mar 1 01:47:49.871: //-1/2B88C4988041/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Mar 1 01:47:50.067: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Calling Number=, Called Number=100, Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 01:47:50.067: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=100
*Mar 1 01:47:50.071: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Result=Partial Matches(1) after DP_MATCH_DEST
*Mar 1 01:47:50.071: //-1/2B88C4988041/DPM/dpMatchPeersMoreArg:
Result=MORE_DIGITS_NEEDED(1)
*Mar 1 01:47:50.267: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Calling Number=, Called Number=1001, Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 01:47:50.271: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=1001
*Mar 1 01:47:50.271: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Mar 1 01:47:50.271: //-1/2B88C4988041/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=10
*Mar 1 01:47:50.283: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=1001, Called Number=1001, Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 01:47:50.287: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=1001
*Mar 1 01:47:50.287: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Mar 1 01:47:50.287: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=10
*Mar 1 01:47:50.299: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Calling Number=, Called Number=1001, Peer Info Type=DIALPEER_INFO_SPEECH
*Mar 1 01:47:50.299: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=1001
*Mar 1 01:47:50.303: //-1/2B88C4988041/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
*Mar 1 01:47:50.303: //-1/2B88C4988041/DPM/dpMatchPeersMoreArg:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=10
Please suggest.
Regards
Amarjit Das
02-19-2014 05:04 AM
from the CME router & the H323 GW, pelase capture 'debug voip ccapi inout' for a test call and let us know the calling/called numbers
//Suresh
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02-19-2014 05:09 AM
Hi Amarjit.
Did you configure any partition for CUCM extensions?
Let me know
Regards
Carlo
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