Cisco3845 ISR CUBE

[markcarat]

Hello

I have a Cisco 3845 router running IOS c3845-advipservicesk9-mz.124-3i.bin and MGPC and two PRIs for PSTN calls.  Now i want to test SIP trunking on this router and TAC told me to upgrade the router to c3845-advipservicesk9-mz.124-24.T6.bin in order to configure CUBE.  I did the upgrade and notice it didn't take some of the command.  I double checked that with Cisco document and looks like I need a different IOS to support feature for CUBE.  The IOS is c3845-adventerprisek9_ivs-mz.124-24.T6.bin.  can anyone shed some light on this?

Also I ran into an issue with translation rule with SIP. Assuming we use 8 to get outside line. We need to set up a rule to strip the 8 from outgoing called number. 

voice translation-rule 8

rule 2 /^8\(.*\)/ /\1/

voice translation-profile DIGITSTRIP-8

translate called 8

However we use * for outgoing call.  So I change the the translation rule to

voice translation-rule 8

rule 2 /^*\(.*\)/ /\1/

but Cisco won't take this command and give me an error. any input would be appreciated.

Thanks

Mark

[boris.kudr · Accepted solution]

Hi Mark,

Please share which commands were not accepted.

You can check feature availability per IOS image here:

http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp

As for the translation, you can use \ symbol to match * in the rule:

voice translation-rule 8

rule 2 /^\*\(.*\)/ /\1/

Cheers,

Boris

[markcarat]

Hi Boris,

Here is the command I got from Cisco TAC

voice class uri Callmanager sip  ---> This one works

host ipv4:x.x.x.x (ip of cucm node #1)  This does not work

host ipv4:y.y.y.y (ip of cucm node #2)  This does not work

And thanks for the translation, this fix the issue

Thanks

Mark

[Chris Deren]

I've configured many CUBEs and never needed to use these.  What are you integrating your CUBE with?  SIP Trunk provider, another PBX?

Chris

[markcarat]

Hi Chris,

I plan to integrate CUBE with a SIP provider.  We have CUCM 8.5 with mgcp gateways and local PRIs.  Ultimate goal is to use SIP trunk and move away from PRIs.  Can the router (CUBE) stay with the current IOS advipservicesk9_mz.124_24.T6?

thanks

Mark

[Gajanan Pande]

Mark,

That should work for you but being with CUCM 8.5, the recommendation would be to go with IOS 15.x to avail full feature set offered by CUCM 8.x - CUBE 8.x combination.

Please refer features documented for CUBE 8.x & also the ordering information for selecting appropriate IOS version for the CUBE, in the datasheet here http://www.cisco.com/en/US/partner/prod/collateral/voicesw/ps6790/gatecont/ps5640/product_data_sheet09186a00801da698.html

HTH

GP.

Pls rate helpful posts !!

[Chris Deren]

I concur, go with 15.X train, i.e. 15.1.4M to get significant CUBE improvements.

Chris

[markcarat]

Thank you GP and Chris,

I ran a test with a SIP provider today.  They gave me test DID to call oer the SIP trunk.  When I made the call, the phone ring but the call would drop once the he picked up the handset. The there is a recording " your call cannot be complete as dial"

Any thoughts

Nark

[Chris Deren]

Can you post your CUBE config and "debug ccsip call", "debug ccsip messages"?

If you installed IOS 15.1, make sure you add the following:

voice service voip

no ip address trusted authenticate

Chris

[markcarat]

Hi CHris,

I have attached the CUBE config and the debug. 

Thanks

Mark

[Chris Deren · Accepted solution]

Looks like your CUBE is sending CANCEL request after about 12 seconds.

Looking at the config couple of things, you need to bind SIP to a particular interface, i.e:

voice service voip

no ip address trusted authenticate

sip

  bind control source-interface Port-channel22

  bind media source-interface Port-channel22

  header-passing error-passthru

  early-offer forced

  midcall-signaling passthru

!

Alos, try the early-media forced as shown above, that is typically required for all SIP providers.

What IOS version did you end up installing?

You are saying the carrier gave you a test number, is it 15852551531?  Becuase the debug shows an outbound call flow not an inboud which is what a test number would be for.

HTH, please rate all useful posts!

Chris

[markcarat]

Hi Chris,

Appreciated your help.  I changed the codec on CM to match the codec on the SIP dial-peer.  The outbound call was connected and the receiving side is able to hear me but I can't hear from the receiving side.  Looks like a one way voice issue at the moment.  The SIP provider told me I need to setup a NAT at the CUBE.  Here is my call flow.

And they said we are requesting them to send the media to IP 10.118.150.5.  How could I set up NAT at CUBE so they will see my public IP address?

SIP provider --> internet --> my firewall --> CUBE --> CUCM

v=0

o=CiscoSystemsSIP-GW-UserAgent 3402 2377 IN IP4 10.118.150.5

s=SIP Call

c=IN IP4 10.118.150.5

t=0 0

m=audio 18886 RTP/AVP 0 101

c=IN IP4 10.118.150.5

Thanks

mark