How do SIP Trunk Configuration

Just make sure the IOS is IP to IP capable, thats the CUBE functionality you need.

Hello,
The IOS is capable , my first question is : Can I configure this router as a CUBE for SIP trink between ITSP and my CUCM at the same time this router working as a H.323 gateway between my CUCM and old AVAYA PBX!  If I can  how?

Well yes, its a matter of dial peers. Whatever pattern you want to treat as SIP and which one as H.323. It seems like this is what you want:

CUCM --> SIP --> CUBE --> SIP --> ITSP
CUCM --> H323 --> CUBE --> H323 --> AVAYA

What exactly do you need assistance or clarification with?

That is nice point : Can I use the first part of second  existing  path to create this scnario:
CUCM --> H323 --> CUBE _______> SIP______>ITSP

or have to create new complete path like what you mentioned above CUCM --> SIP --> CUBE --> SIP --> ITSP .

we did this before it will be working fine but you have to consider the fax issues.

now you can keep your H323 gateway and create SIP dial-peers to your ITSP so it will be as you mentioned above:
CUCM --> H323 --> CUBE _______> SIP______>ITSP

but be sure that you dont have to register with the ITSP otherwise you have to configure
sip-ua

Thanks

If I already register with ITSP for 2000 DID range  for xxxx1000 to xxxx2999  please advise me how the dial-peers looks like?

dial-peer voice 1 voip
destination-pattern ...... (the number that you want to dial)
session protocol sipv2
session target ipv4:x.x.x.x (your ITSP IP Address)
codec (use the correct codec)
no vad
dtmf relay (use either sip-notify or rtp-nte)

I think your provider should see the calling number is one of the DID so use full number from the call manager

Can I use just one destination pattern for all out going call s( local- mobile international) like .T  or have to use one dial-peer for each?   What about coming calls from out side to IP phones behind CUCM.?

yes you can use one dial-peer but .T is not recommended use 9T as 9 is your access code

the dial peer going to CUCM will be

dial-peer voice 2 voip
destination-pattern .... (your extensions)
session target ipv4:x.x.x.x (CUCM IP Address)
codec (the codec that you want to use between the call manager and the gateway, you can use voice class if you want)
no vad
dtmf relay h245-alphanumeric


dont forget to configure the following for sip and h323 interconnection

voice service voip
allow-connection h323 to sip
 allow-connection h323 to h323
allow-connection sip to sip
allow-connection sip to h323

Thanks a lot for you huns1984 , It seems good for now ! I will do it upon the SIP trunk and DID become ready during 1 or 2 weeks and I will tell you the result.  

waiting for you.

I'm so sorry but I think I have one point still missing. now suppose that call coming from out side where some body dial xxxx2016 as example how the cucm forward this call to the extension 2016 ? what the configuration required on the cucm side for DID? Is it new Rule have to applied ?

no need to set any rule just go to the gateway page and set the significant digits to 4

Hello there,
Sip service now is up and the statistic and status command output like in attached. but call can't establish properly. any advice?  
sip-statistic.txt
sip-stat.txt

do you have to register your gateway to your provider? or you have to send the call to a specific IP?

I'm not sure what u mean! any way I have no applied any registration configuration in my gateway the provider only give us sip Ip address .

what codec they are using?

they use  G711A-Law

and they use DTMF in band with RFC2833

ok so as I mentioned before you can send them the call using the following dial-peer:

dial-peer voice 1 voip
destination-pattern ...... (the number that you want to dial)
session protocol sipv2
session target ipv4:x.x.x.x (your ITSP IP Address)
codec g711a
no vad
dtmf relay rtp-nte

test it and tell me

I already made the same configuration with destination-pattern  9T  ( the number that I need dial from 9 digits ). the coming tune after dial like network busy

aha OK, you have to create one translation to remove the 9. use the following:

voice translation-rule 1
rule 1 /^9/ //

voice translation-profile SIP-OUT
translate called 1

dial-peer vocie (your dial-peer)
translation-profile outgoing SIP-OUT

after this it should work.

I did it with nothing!  see in call manager we already check discard PreDot .
I think there is somthing error in the configuration in call manager not in gateway!

OK when you have 9.XXXXX with predot the call is reaching the gateway without 9 so you dont have dial-peer that match this number so set the discard digits to none it should work, if not please do debug voice dial in and make a call then post the output

ok I applied this rule on gateway but still nothing! any way I think there is other issue with the provide   let them reply me then I will update you . It will be tomorrow

ok waiting for you

Hell there,
they provided me with new sip ip address but the situation now like this! when I call an extention from mobile . extention is ringing but can't recive the call .

you mean outgoing calls are working fine but incoming calls not?

I'm sorry
the first thing there was technical problem with provider and they gave us new SIP ip address. then from my side all outgoing calls were dropped and incoming call just keep ringing even after hangup.
any way at the end I add transcoder and MTP configuration like below and every thing is ok now so thanks very much for you and for ex.

sccp local GigabitEthernet0/1

sccp ccm x.x.x.x identifier 1 priority 1 version 7.1    ( ccm ip add)
sccp
!
sccp ccm group 1
 bind interface GigabitEthernet0/1
 associate ccm 1 priority 1
 associate profile 1 register XCD123456
 associate profile 2 register MTP123456
!
dspfarm profile 1 transcode  
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 8
 associate application SCCP
!
dspfarm profile 2 mtp  
 codec g711ulaw
 maximum sessions hardware 8
 maximum sessions software 20
 associate application SCCP
!

then configure hard ware mtp and transcoder in ccm.


note: I don't know why destination pattern (9T) in outgoing dial peer not working so I replaced it by this one (.T) .

why do you need mtp and xcod?
.T is not recomended as it can match any thing.

but if you are ok with this solution, keep it as is

mtp and xcod because I have different codding and different protocols.

It is great info. source and good location for sharing information.