SIP translation rule

[Sulthanfathim]

Hello Experts,

ITSP has provided 10 SIP extension lines, and lines are capable of both incoming and outgoing features too. we have around 45 IP phones and everyone needs to call outside using destination Pattern 9T

how to configure translation rule for 10 sip outgoing lines. our intention is everyone will use a 9T pattern strip. if any line SIP line is busy, the system automatically checks the vacant sip line and route to that line.

example :

one of IP phone user EXT 100 is calling to outside over 9T (905554493163) and his call is being connected.

At the same time another IP phone user EXT 101 needs to call outside another using 9T, so the system should be analyzed SIP line 1 is busy and traffic should be route into SIP line 2 . like that up to 10 users 

Now i applied only one rule when a call over 9T translation rule will transfer to SIP Line 1 which number is 5404915

voice translation-rule 3

rule 1 /.*/ /5404915

when the above outgoing line is busy.i need to configure the remaining below numbers to outgoing call over 9T 

5404910
5404911
5404912
5404913
5404914
5404915
5404916
5404917
5404918
5404919

current rule as below

voice translation-rule 1

rule 1 /0115404910/ /500/

rule 2 /0115404911/ /500/

rule 3 /0115404912/ /500/

rule 4 /0115404913/ /500/

rule 5 /0115404914/ /500/

rule 6 /0115404915/ /500/

rule 7 /0115404916/ /500/

rule 8 /0115404917/ /500/

rule 9 /0115404918/ /500/

rule 10 /0115404919/ /500/

!

voice translation-rule 2

rule 1 /^9\(.*\)/ /\1/

!

voice translation-rule 3

rule 1 /.*/ /5404915

dial-peer voice 100 voip

description **Incoming -Outgoing Call from SIP Trunk**

translation-profile incoming CUE_AutoAttendant

translation-profile outgoing SIP_Outgoing

destination-pattern 9T

session protocol sipv2

session target ipv4:10.141.40.233:5060

session transport udp

incoming called-number .

voice-class codec 1 

voice-class sip bind control source-interface GigabitEthernet0/1

voice-class sip bind media source-interface GigabitEthernet0/1

dtmf-relay rtp-nte

no vad   

[Chris Deren]

I am confused, are you having issues with outbound or inbound calls?

You state that when you send the call out with 9 it works for one user but not for the other? Are they dialing different external number?

Normally you would want to just strip the offset access code on outbound calls i.e. 9 and you can do that by simply using this translation rule/profile:

voice translation-rule 9

rule 1 /^9/ //

voice translation-profile strip-9
translate called 9

and apply it to the outbound dial-peer:

dial-peer voice 100 voip

translation-profile outgoing strip-9

let me know if I missed anything, and if this does not work, please provide full "show run", "debug ccsip messages" and "debug voice translation" along with calling and called numbers.

[Sulthanfathim]

Thanks for your respond,

Based on the Rule#1, All SIP (10 numbers) line incoming calls are perfectly hiting in CUE AA pilot 500 Extension. only the problem Outbound call not euqlly distributing to other SIP lines

*****************************************************

based on your suggesion configuration has changed as below

voice translation-rule 1
rule 1 /0115404910/ /500/
rule 2 /0115404911/ /500/
rule 3 /0115404912/ /500/
rule 4 /0115404913/ /500/
rule 5 /0115404914/ /500/
rule 6 /0115404915/ /500/
rule 7 /0115404916/ /500/
rule 8 /0115404917/ /500/
rule 9 /0115404918/ /500/
rule 10 /0115404919/ /500/
!
voice translation-rule 3
rule 1 /^9/ //
!
!
voice translation-profile CUE_AutoAttendant
translate called 1
!
voice translation-profile SIP_Outgoing
translate called 3

dial-peer voice 100 voip
description **Incoming -Outgoing Call from SIP Trunk**
translation-profile incoming CUE_AutoAttendant
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target ipv4:10.141.40.233:5060
session transport udp
incoming called-number .
voice-class codec 1 
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
no vad

******************************************************************

Above configuration i can't make any out going calls from IP phones its getting a long busy tone, however when i make below configuration every IP Phone can able to make outgoing calls over 9 strip and end point mobile phonecaller ID displaying rule#3 SIP #0115404910 Too

voice translation-rule 1
rule 1 /0115404910/ /500/
rule 2 /0115404911/ /500/
rule 3 /0115404912/ /500/
rule 4 /0115404913/ /500/
rule 5 /0115404914/ /500/
rule 6 /0115404915/ /500/
rule 7 /0115404916/ /500/
rule 8 /0115404917/ /500/
rule 9 /0115404918/ /500/
rule 10 /0115404919/ /500/
!
voice translation-rule 2
rule 1 /^9\(.*\)/ /\1/
!
voice translation-rule 3
rule 1 /.*/ /5404910/
!
!
voice translation-profile CUE_AutoAttendant
translate called 1
!
voice translation-profile SIP_Outgoing
translate calling 3
translate called 2

But the Problem now,  all calls are going to only one SIP line#(5404910), because, rule#3 clearly mentioned choose the line as below

voice translation-rule 3
rule 1 /.*/ /5404910/

Since ITSP provided 10 SIP line numbers, if one of the IP phone user is calling to outgoing using By 9T strip, the same time other IP phone users also need to call outside using 9T strip. so the second user calls automatically route into the other SIP lines, this is our Basic Intension 

Please refer debug call logs which is previous and as you suggested both configurations

NB : I'm a beginner in Cisco Voice spectrum, Give me more details as  crispy, how to Digg deeply

With best regards

Sulthan

[Chris Deren · Accepted solution]

With SIP trunking there is not "lines", you have number of concurrently allowed calls per the SIP trunk.  

Let's separate inbound from outbound calls and tackle them separately and for now focus on outbound call.

The call that did not work failed with reason "SIP/2.0 484 Address Incomplete", if you compare the working vs. not working call you will see that the non-working call presented the caller ID as 3 digit (199) and telco did not like it.  So, you need to mask your ANI (caller ID) on outbound calls.  The question is what do you want to mask it to? Do you want to use one main number for all calls going out or you want each phone to be masked with different caller ID?  I think you were trying to figure out away to mask the caller ID for each call to different number, which cannot be done easily nor should be the goal.  Since all inbound calls go to AA anyway, why not just maks all outbound calls with one selected number (call it your main number)?  

Your current config of:

voice translation-profile SIP_Outgoing
translate calling 3
translate called 2

should mask all outbound calls with ANI=5404910

So, all outbound calls should mask the caller ID with the above number, and you should be able to make 10 concurrent inbound calls to this 1 number.

[Sulthanfathim]

Spoiler

With SIP trunking there is not "lines", you have number of concurrently allowed calls per the SIP trunk.

Thanks for your above information.

[Chris Deren · Accepted solution]

Excellent, if you do not have any further questions please remember to rate all useful posts to help others in the future. 

[Sulthanfathim]

Hello Mr. Chris Deren

Thanks for your assistance, regarding SIP Trunk concurrent concept.

i have a simple question too.

we have a PSTN analog telephone line which is our main number (0115405171) it is connected FXO voice-port 0/0/0

Also, we have SIP Lines ( 0115404910-19) , Based on below rule now all calls are being routed over Sip trunk which is the called ID 5404919

voice translation-rule 3
rule 1 /.*/ /5404919/

Is there any way to set a caller ID which is the main number 0115405171(Analog number) for all sip calls ?

i.e.

whenever call going to outside network over Sip trunk, the caller ID displayed on the end-user mobile screen as our main number (0115405171)

************************************************************************************************

curretn configuration as below :

voice translation-rule 1
rule 1 /0115404910/ /500/
rule 2 /0115404911/ /500/
rule 3 /0115404912/ /500/
rule 4 /0115404913/ /500/
rule 5 /0115404914/ /500/
rule 6 /0115404915/ /500/
rule 7 /0115404916/ /500/
rule 8 /0115404917/ /500/
rule 9 /0115404918/ /500/
rule 10 /0115404919/ /500/
!
voice translation-rule 2
rule 1 /^0\(.*\)/ /\1/
!
voice translation-rule 3
rule 1 /.*/ /5404919/
!
!
voice translation-profile CUE_AutoAttendant
translate called 1
!
voice translation-profile SIP_Outgoing
translate calling 3
translate called 2
!

*********************************************************

[Chris Deren · Accepted solution]

You can try to translate it to the main number, but some SIP providers will reject calls unless you send it with caller id of DID from the SIP trunk, for that you could try diversion header. So, here is what to try:

Change

voice translation-rule 3
rule 1 /.*/ /5404919/

to

voice translation-rule 3
rule 1 /.*/ /5405171/

test if it works, if not add the following:

voice class sip-profiles 1
request INVITE sip-header Diversion modify "<sip:(.*>)" "< sip:5404919@10.170.13.126>"

dial-peer voice 100 voip

voice-class sip profiles 1

Keep in mind though that if you present caller ID of your FXO trunks, people may call it instead of your SIP number.  You can also port your FXO numbers to your SIP trunk by arranging that with your provider.

[Sulthanfathim]

Once Again Thanks for your support,

As suggested,I try to change as below, however, call get rejected, even added the header,

**************************************************

voice class sip-profiles 1

  request INVITE sip-header Diversion modify "<sip:(.*>)" "< sip:5404919@10.170.13.126>"

voice translation-rule 2
rule 1 /^0\(.*\)/ /\1/
!
voice translation-rule 3
rule 1 /.*/ /5405171/                                          // call not going when i put FXO Number
!
!
voice translation-profile CUE_AutoAttendant
translate called 1
!
voice translation-profile SIP_Outgoing
translate calling 3
translate called 2
!

dial-peer voice 100 voip
description **Incoming -Outgoing Call from SIP Trunk**
translation-profile incoming CUE_AutoAttendant
translation-profile outgoing SIP_Outgoing
destination-pattern 0T
session protocol sipv2
session target ipv4:10.141.40.233:5060
session transport udp
incoming called-number .
voice-class codec 1 
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
no vad

********************************************************************

The call can connect only when adding SIP number in rule#3 as below

voice translation-rule 3
rule 1 /.*/ /5404919/

Is there any other suggestion or solutions?

[Chris Deren]

Please attach "debug ccsip messages"

[Sulthanfathim]

Please refer attachment

[Chris Deren · Accepted solution]

can you add the following

voice class sip-profiles 1

request INVITE sip-header Diversion add "<sip:5404919@10.170.13.126>"

so it looks like:

voice class sip-profiles 1
request INVITE sip-header Diversion modify "<sip:(.*>)" "< sip:5404919@10.170.13.126>"

request INVITE sip-header Diversion add "<sip:5404919@10.170.13.126>"

provide same debug.

[Sulthanfathim]

Please refer attachment debug result

I try to modify the header as suggested but outgoing calls failure