Call Flows and Call Legs in UCM and UCME
Provisioning Basic Setup:
--------------------------------
Note: We would not use the publisher as call processing node in a production environment.
Note: You could issue the subscriber as the primary processing node and the publisher as a backup
ON R1:
--------------
show ip dhcp binding
conf t
ip dhcp pool CME-CorHQ-PHONES
no option 150 ip 177.1.254.1
option 150 ip 177.1.10.10 177.1.10.20
ON R2:
---------------
conf t
ip dhcp pool CME-Branch1-PHONES
no option 150 ip 177.1.254.1
option 150 ip 177.1.10.10 177.1.10.20
NOW SSH TO THE PUBLISHER:
-----------------------------------------
ssh -l admin 177.1.10.11
NOW SSH TO THE SUBSCRIBER:
--------------------------------------------
ssh -l admin 177.1.10.12
show myself
utils system
If the Phone MAC starts with the following then it is a SCCP.
SEP.....
And SIP Phones can load MGCP and H.323 protocol
AND BETWEEN the Unity Express and the CUCME the protocol is only SIP
In case it is integrated with the Call Manager it talks JTAPI 'Java Telephony Application Programming Interface'
AIM-CUE 'CF Flash-based storage'
ISM-SRE-300
ISM-SRE-700
TO restart the CM:
ssh -l 177.1.10.11 admin password
utils system restart
TO role back:
utils system switch-version
TO change the password of your account:
set account
AAR ' Automated Alternate Route:
This is like the PSTN Backup but in this case it is in the WAN, but when there is a lack of bandwidth
Calling = ANI 'Automatic number identification'
Called = DNIS 'Dialed Number Identification Service'
Redirec Called = R-DNIS
Note:
e.g.
--------
dial-peer voice 10 pots
destination-pattern 911$
port 0/0/0:23
forward-digits 3
----------
If the type of the dial peer is pots then the dial peer by default strips any explicity matches digit.
Anywhere in the world YOU will have the following forms of dial-peers:
1-Emergency Services Numbers:
dial-peer voice 10 pots
destination-pattern 911$
port 0/0/0:23
forward-digits 3
2-Local Number:
dial-peer voice 11 pots
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
forward-digit 10
3-National or Long distance:
dial-peer voice 12 pots
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digit 11
4-International Number:
dial-peer voice 13 pots
destination-pattern 9011T
port 0/0/0:23
prefix 011
For analog calls that going through fxo we will point it to the port 0/0/0 'NOTE without the :23'
IF I have 4 ports to the PSTN: I would have to dial out 4 dial peers
1-one for Emergency Services Numbers.
2-one for Local Number.
3-one for National or Long distance
4-and one International Number
and will have the above related configurations for 4 times and changing the port number from
0/0/0
0/0/1
0/0/2
0/0/3
Or best way is to create a Trunk Group
So the FXO ports is considered a TDM. so the TDM circuit is called Trunk.
and then after creating the trunk group we assocaite the port to the trunk and then the dial peer instead of writing 4 times of each one we write only one and instead of referencing port number we reference the trunk group
#When dealing with FXS port you might say how get a call to ring out to that particular analog phone that is connected to it.
It is actually the exact same way as the outgoing FXO port we still have a destination-pattern BUT in the FXS port we will never have a variable numbers e.g. 2[2-9]..[2-9]......$, so variable numbers only when I am sending variable number on the PSTN or onto another PBX, so variable number never go to the analog phones because they only understand ring and talk.
So the analog phone does not understand:
-Call-hold
-Call transfere
-Call forward
-Message waiting indicator
-Additional Digits
Only the GW do everything and then send only :
Caller id or Calling Number
#Note Whenever we have overlaping we do not write the $ sign in the destination-pattern, e.g.
destination-pattern 91[2-9]..[2-9].....$
Note: When we have those ^24242...$ e.g. ^ and $ we mean only this amount of numbers will be there no more than that.
#With Voip dial-peers
#With the command:
dial-peer hunt 'we can set the preference of the calling number based on the following matches:
-Logest match
-Explicit preference
-Random selection
-Least recent use.
So dial-peer hunting is related to the outbound dial-peers search only,
and we can stop it from hunting and not to go to the next one using this command in the dial-peer :
stophunting
#Inorder to match to the inbound call leg:
We need to specify based on one of four options which go in the following order:
1-dial-peer voice 1 pots
incoming called-number .
2-dial-peer voice 1 pots
answer-address
3-dial-peer voice 1 pots
destination-pattern
4-dial-peer voice 1 pots
port
#Note :
-If the above options are pots dial-peers then this is for digital trunks.
so for this type of dialpeers e.g.
T1/E1 CAS or T1/E1 PRI or BRI then we have the option of matching based on the inbound digits so the first choice is :
First: 'incoming called-number .'
and the incoming number means the 'Called Number = DNIS'
with the use of '.' at the end which is enough to match any on all calls.
Second: Is if we have 'answer-address .'
which is 'Calling Number = ANI'
Third: destination-pattern:
Which actually matches 'Calling Number = ANI'
On outbound the destination-pattern matches 'Called Number = DNIS', and On inbound the destination-pattern mateches 'Calling Number = ANI'
The three above only take effect if the type is Voip dial peer.
Four: Port is only gonna matches if the dial-peer type is pots
-BUT if the call is coming from analog trunk through FXS well then it is going to match whatever was sent to it in terms of the call digit that is going to be dialed into the system.
-But if the call is coming from FXO port we are simply going to at the voice-port level
e.g.
voice-port 0/1/3
connection-plar 3000
So the connection plar says whenever an FXO call or an FXS port call comes in, THEN we are going to send it to specific destination
#NOTE:
in the following dial peer we have the command 'direct-inward-dial'
dial-peer voice 1 pots
incoming called-number
direct-inward-dial
it means take the digits that just received this dial peer and pass them on to the rest of the dial peers for matching, if we do not have it then, the call comes in it matches the first dial-peers because of the 'incoming called-number . and the it places the dial tone and the router just waits or someone dial a number because I did not pass the digits that already passed to me onto the rest of the dial-peers.
#The two dial peers that for the publisher and subscriber:
-The publisher dial peers:
-----------------
dial-peer vouce 101 voip
translation-profile incoming Prefix_9_DNIS
preference 1
destination-pattern 5126022...
session target ipv4:177.1.10.11
incoming called number .
voice-class codec 1
voice-class h323 1
dtmf-relay h245-signal
-------------------
Note the 'incoming called number .' is here because it is the publisher for the VOIP
-The subscriber dial peers:
------------------
dial-peer voice 102 voip
translation-profile incoming Prefix_9_DNIS
preference 2
destination-pattern 5126022...
session target ipv4:177.1.10.12
voice-class codec 1
voice-class h323 1
dtmf-relay h245 1
dtmf-relay h245-signal
Provisioning Basic Setup:
--------------------------------
Note: We would not use the publisher as call processing node in a production environment.
Note: You could issue the subscriber as the primary processing node and the publisher as a backup
ON R1:
--------------
show ip dhcp binding
conf t
ip dhcp pool CME-CorHQ-PHONES
no option 150 ip 177.1.254.1
option 150 ip 177.1.10.10 177.1.10.20
ON R2:
---------------
conf t
ip dhcp pool CME-Branch1-PHONES
no option 150 ip 177.1.254.1
option 150 ip 177.1.10.10 177.1.10.20
NOW SSH TO THE PUBLISHER:
-----------------------------------------
ssh -l admin 177.1.10.11
NOW SSH TO THE SUBSCRIBER:
--------------------------------------------
ssh -l admin 177.1.10.12
show myself
utils system
If the Phone MAC starts with the following then it is a SCCP.
SEP.....
And SIP Phones can load MGCP and H.323 protocol
AND BETWEEN the Unity Express and the CUCME the protocol is only SIP
In case it is integrated with the Call Manager it talks JTAPI 'Java Telephony Application Programming Interface'
AIM-CUE 'CF Flash-based storage'
ISM-SRE-300
ISM-SRE-700
TO restart the CM:
ssh -l 177.1.10.11 admin password
utils system restart
TO role back:
utils system switch-version
TO change the password of your account:
set account
AAR ' Automated Alternate Route:
This is like the PSTN Backup but in this case it is in the WAN, but when there is a lack of bandwidth
Calling = ANI 'Automatic number identification'
Called = DNIS 'Dialed Number Identification Service'
Redirec Called = R-DNIS
Note:
e.g.
--------
dial-peer voice 10 pots
destination-pattern 911$
port 0/0/0:23
forward-digits 3
----------
If the type of the dial peer is pots then the dial peer by default strips any explicity matches digit.
Anywhere in the world YOU will have the following forms of dial-peers:
1-Emergency Services Numbers:
dial-peer voice 10 pots
destination-pattern 911$
port 0/0/0:23
forward-digits 3
2-Local Number:
dial-peer voice 11 pots
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
forward-digit 10
3-National or Long distance:
dial-peer voice 12 pots
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digit 11
4-International Number:
dial-peer voice 13 pots
destination-pattern 9011T
port 0/0/0:23
prefix 011
For analog calls that going through fxo we will point it to the port 0/0/0 'NOTE without the :23'
IF I have 4 ports to the PSTN: I would have to dial out 4 dial peers
1-one for Emergency Services Numbers.
2-one for Local Number.
3-one for National or Long distance
4-and one International Number
and will have the above related configurations for 4 times and changing the port number from
0/0/0
0/0/1
0/0/2
0/0/3
Or best way is to create a Trunk Group
So the FXO ports is considered a TDM. so the TDM circuit is called Trunk.
and then after creating the trunk group we assocaite the port to the trunk and then the dial peer instead of writing 4 times of each one we write only one and instead of referencing port number we reference the trunk group
#When dealing with FXS port you might say how get a call to ring out to that particular analog phone that is connected to it.
It is actually the exact same way as the outgoing FXO port we still have a destination-pattern BUT in the FXS port we will never have a variable numbers e.g. 2[2-9]..[2-9]......$, so variable numbers only when I am sending variable number on the PSTN or onto another PBX, so variable number never go to the analog phones because they only understand ring and talk.
So the analog phone does not understand:
-Call-hold
-Call transfere
-Call forward
-Message waiting indicator
-Additional Digits
Only the GW do everything and then send only :
Caller id or Calling Number
#Note Whenever we have overlaping we do not write the $ sign in the destination-pattern, e.g.
destination-pattern 91[2-9]..[2-9].....$
Note: When we have those ^24242...$ e.g. ^ and $ we mean only this amount of numbers will be there no more than that.
#With Voip dial-peers
#With the command:
dial-peer hunt 'we can set the preference of the calling number based on the following matches:
-Logest match
-Explicit preference
-Random selection
-Least recent use.
So dial-peer hunting is related to the outbound dial-peers search only,
and we can stop it from hunting and not to go to the next one using this command in the dial-peer :
stophunting
#Inorder to match to the inbound call leg:
We need to specify based on one of four options which go in the following order:
1-dial-peer voice 1 pots
incoming called-number .
2-dial-peer voice 1 pots
answer-address
3-dial-peer voice 1 pots
destination-pattern
4-dial-peer voice 1 pots
port
#Note :
-If the above options are pots dial-peers then this is for digital trunks.
so for this type of dialpeers e.g.
T1/E1 CAS or T1/E1 PRI or BRI then we have the option of matching based on the inbound digits so the first choice is :
First: 'incoming called-number .'
and the incoming number means the 'Called Number = DNIS'
with the use of '.' at the end which is enough to match any on all calls.
Second: Is if we have 'answer-address .'
which is 'Calling Number = ANI'
Third: destination-pattern:
Which actually matches 'Calling Number = ANI'
On outbound the destination-pattern matches 'Called Number = DNIS', and On inbound the destination-pattern mateches 'Calling Number = ANI'
The three above only take effect if the type is Voip dial peer.
Four: Port is only gonna matches if the dial-peer type is pots
-BUT if the call is coming from analog trunk through FXS well then it is going to match whatever was sent to it in terms of the call digit that is going to be dialed into the system.
-But if the call is coming from FXO port we are simply going to at the voice-port level
e.g.
voice-port 0/1/3
connection-plar 3000
So the connection plar says whenever an FXO call or an FXS port call comes in, THEN we are going to send it to specific destination
#NOTE:
in the following dial peer we have the command 'direct-inward-dial'
dial-peer voice 1 pots
incoming called-number
direct-inward-dial
it means take the digits that just received this dial peer and pass them on to the rest of the dial peers for matching, if we do not have it then, the call comes in it matches the first dial-peers because of the 'incoming called-number . and the it places the dial tone and the router just waits or someone dial a number because I did not pass the digits that already passed to me onto the rest of the dial-peers.
#The two dial peers that for the publisher and subscriber:
-The publisher dial peers:
-----------------
dial-peer vouce 101 voip
translation-profile incoming Prefix_9_DNIS
preference 1
destination-pattern 5126022...
session target ipv4:177.1.10.11
incoming called number .
voice-class codec 1
voice-class h323 1
dtmf-relay h245-signal
-------------------
Note the 'incoming called number .' is here because it is the publisher for the VOIP
-The subscriber dial peers:
------------------
dial-peer voice 102 voip
translation-profile incoming Prefix_9_DNIS
preference 2
destination-pattern 5126022...
session target ipv4:177.1.10.12
voice-class codec 1
voice-class h323 1
dtmf-relay h245 1
dtmf-relay h245-signal
No comments:
Post a Comment