Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
Announcements

Welcome to Cisco Support Community. We would love to have your feedback.

For an introduction to the new site, click here. And see here for current known issues.

New Member

Multiple SIP Lines Design Requirment

Current Configuration:

ip route 10.230.0.26 255.255.255.255 10.240.23.233

interface GigabitEthernet0/1.1
 encapsulation dot1Q 1 native
 ip address 10.240.23.234 255.255.255.248
 ip traffic-export apply CAPTURE size 10000000

dial-peer voice 3 voip
description OUTGOING SIP Line for Association-A
destination-pattern *9.......
 no modem passthrough
 session protocol sipv2
 session target ipv4:10.230.0.26
 voice-class codec 1 

 

This is the configuration of having one SIP line, on our interface having IP 10.240.23.234 and on SIP service provider end having 10.240.23.233.

Whereas SIP Server IP remains the same as said by SIP Service Provider, i.e 10.230.0.26, which is used by in our dial-peers,

 

Question: One more association is added to the building which want to use different SIP line for calls. No problem I can make different dial-peer for each association-A and association-B as shown below, Where as I can remove 8 after the dial peer 4 is match or when the call is going out, but in both dial-peer I am using the same SIP server i.e 10.230.0.26, so weather a call is coming from association-A or association-B it will direct calls to SIP server IP, so as per the static route we said all packets having destination of SIP server (10.230.0.26) should go to 10.240.23.233 and If we configure for second SIP line we can add this static route i.e

ip route 10.230.0.26 255.255.255.255 10.240.22.2,

So now two route exist for same destination, can anyone give me some hint on it what can be done, how association-A can use association-A line for calls and association-B by using association-B line, I am using SIP trunk between Voice Gateway 2911 and CUCM 8.6.2

I have to give design to management that it can work or not, I am just stuck at this point only, not sure what to do, thank you.

 

like,

 

Altered Configuration:

 

interface GigabitEthernet0/1.1
 encapsulation dot1Q 1 native
 ip address 10.240.23.234 255.255.255.248
 ip traffic-export apply CAPTURE size 10000000

interface GigabitEthernet0/1.2
 encapsulation dot1Q 1 native
 ip address 10.240.22.1 255.255.255.248
 ip traffic-export apply CAPTURE size 10000000

 

ip route 10.230.0.26 255.255.255.255 10.240.23.233

ip route 10.230.0.26 255.255.255.255 10.240.22.2

 

dial-peer voice 3 voip
description OUTGOING SIP Line for Association-A
destination-pattern *9.......
 session protocol sipv2
 session target ipv4:10.230.0.26
 voice-class codec 1 

dial-peer voice 4 voip
description OUTGOING SIP Line for Association-B
destination-pattern *89.......
 session protocol sipv2
 session target ipv4:10.230.0.26
 voice-class codec 1 

 

 

PFA the diagram, thank you.

 

--

Kind Regards,

Waleed khan

Everyone's tags (1)
6 REPLIES
New Member

Hi All, Is it possible to

Hi All,

 

Is it possible to solve this by using NAT at SIP service provider. I mean I'll use different IP under my dial-peer for different association. Like,

dial-peer voice 3 voip
description OUTGOING SIP Line for Association-A
destination-pattern *9.......
 session protocol sipv2
 session target ipv4:1.1.1.1
 voice-class codec 1 

dial-peer voice 4 voip
description OUTGOING SIP Line for Association-B
destination-pattern *89.......
 session protocol sipv2
 session target ipv4:2.2.2.2
 voice-class codec 1

 

and

 

ip route 1.1.1.1 255.255.255.255 10.240.23.233

ip route 2.2.2.2 255.255.255.255 10.240.22.2

 

So is it possible at SIP service Provider that they will do NAT and traffic for 1.1.1.1 will be NATED to

10.230.0.26 and for 2.2.2.2 will be NATED to 10.230.0.26.

 

--

Kind Regards,

Waleed Khan

New Member

Dear All,Can someone please

Dear All,

Can someone please look into this issue for me, thank you.

 

--

Kind Regards,

Waleed Khan

dial-peer voice 8863 voip

dial-peer voice 8863 voip
 corlist outgoing PT-Intl
 description SIP to ///Cell ///01.. ///02..
 translation-profile outgoing ///cell_out_sip///
 preference ..
 max-conn .5
 no shutdown
 destination-pattern ///7........
 session protocol sipv2
 session target ipv4://5./7.//5./7
 voice-class sip bind control source-interface GigabitEthernet0/0.310
 voice-class sip bind media source-interface GigabitEthernet0/0.310

New Member

Dear Tagir, Thank you for

Dear Tagir,

 

Thank you for reply, can you please contribute here more by explaining please.

 

- Waleed

New Member

Do I have to use CUBE, can

Do I have to use CUBE, can you please explain if you can contribute here, thanks.

 

- Waleed

New Member

http://www.cisco.com/c/en/us

http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-border-element/white_paper_c11-613550.html?cachemode=refresh

 

I just get to it, let see, if this works or not.

101
Views
3
Helpful
6
Replies
CreatePlease login to create content