cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
1724
Views
5
Helpful
9
Replies

Switching to SIP

sprintership
Level 1
Level 1

Hello,

 

Currently I do have voice gateway Cisco 2900 and Call manager 10.5 publisher  and subscriber.

Couple of questions:

 

1. For firewall rules SIP & RTP - do I allow voice gateway IP for these ports or subscriber|publisher? 

2. My understanding is SIP trunk is set up on call manager or voice gateway?

3. I guess i need some license for SIP on my voice gateway Cisco, I do have this:

 

Cisco CISCO2951/K9 (revision 1.1) with 479232K/45056K bytes of memory.
Processor board ID 
3 Gigabit Ethernet interfaces
24 Serial interfaces
1 terminal line
1 Channelized T1/PRI port
4 Voice FXO interfaces
8 Voice FXS interfaces
DRAM configuration is 72 bits wide with parity enabled.
255K bytes of non-volatile configuration memory.
255488K bytes of ATA System CompactFlash 0 (Read/Write)


License Info:

License UDI:

-------------------------------------------------
Device# PID SN
-------------------------------------------------
*1 CISCO2951/K9 

 

Technology Package License Information for Module:'c2951'

------------------------------------------------------------------------
Technology Technology-package Technology-package
Current Type Next reboot
------------------------------------------------------------------------
ipbase ipbasek9 Permanent ipbasek9
security None None None
uc uck9 Permanent uck9
data None None None
NtwkEss None None None
CollabPro None None None

Configuration register is 0x2102

 

Any help appreciate. Thank you

9 Replies 9

Ritesh Desai
Spotlight
Spotlight
Hi,


1. If firewall is between 2900 voice Router and Cucm pub and subscriber, in that case you require to open firewall ports. You can search on google as “CUCM port utilization guide v1X.y
2. You will need to create SIP trunk between cucm and router. On voice router create dial peer with sip having cucm as session target. If you plan to have ITSP SIP trunk then you need to contact telco who can provide you SIP trunk.
3. For Voice Router to function as SIP you need to remove FXO/FXS / PRI configurations and enable SIP configurations. Cisco do not recommend and TAC do not support. In case you buy sip trunk service from telco then you require CUBE-STD trunk licenses. 1 call 1 license. Basis on Router model, the license capacity is limited. Since you are having very old model, need to check if Cisco provides CUBE-STD trunk licenses for 2900 router.

Hope this helps...
*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

3. For Voice Router to function as SIP you need to remove FXO/FXS / PRI configurations and enable SIP configurations. Cisco do not recommend and TAC do not support.

 

No, that's wrong. Any ISR can have MGCP, H.323, SIP and SCCP all at the same time. What you're talking about is ONLY when you are going to configure CUBE HA, THEN you should not have any TDM alongside with CUBE.

HTH

java

if this helps, please rate

Thanks @Jaime Valencia for clarification +5

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

1. If firewall is between 2900 voice Router and Cucm pub and subscriber, in that case you require to open firewall ports. You can search on google as “CUCM port utilization guide v1X.y

 

There is no firewall between Cisco 2900 and Call manager. 


2. You will need to create SIP trunk between cucm and router. On voice router create dial peer with sip having cucm as session target. If you plan to have ITSP SIP trunk then you need to contact telco who can provide you SIP trunk.

 

I already have SIP trunk provider. I dont understand, So I guess I need two trunks: 1. between CCUM and voice gateway 

2. On the voice gateway SIP trunk to SIP provider - is that correct? 

Yes that would be correct if you use SIP as the control protocol for your voice gateway in CUCM. Think of it as your voice gateway has two sides. Side A is pointing towards CUCM and side B is pointing towards the ITSP for the SIP trunk to PSTN.



Response Signature


I allow only SIP port (can be whatever between ISP SIP provider) and RTP ports only from & to voice gateway which is my CUBE, no need to expose anything from the CCUM. 

 

 

That is also correct. The voice gateway act as a Session Border Controller (SBC) to create a boundary between your corporate network and the service provider. In the SBC you’d want to have an ACL setup and attached to the interface towards the SP that only allows the needed traffic.



Response Signature


Thank you all for all of the explanation. 

@sprintership  if you wish to have SIP trunk from service provider then you can follow the article published by me here  what would SIP parameters would be required for planning.

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai