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SIP trunk

we have a new SIP line from PSTN of bandwidth 16mb(not sure about the bandwidth)the ISP given one access netwotk 124 modem, i need to terminate that line to our H.323 voice gateway ,can anyone please tell me about what configuration i need to do in my VOICE GATEWAY ,we are using CUCM 6.0.1,and i need to do anything this modem .....


Posted by WebUser Jiju Payannur from Cisco Support Community App

Hall of Fame Super Silver

SIP trunk

What is 124 modem?

For SIP trunk termination it is recommended to deploy a CUBE which is a IOS configuration on top of Cisco router, you can reuse your H323 GW and simply configure it for CUBE, remember you need CUBE licenses to run it legally.

As to configuration it varies based on ISP provider and Cisco has some recommendations for major carriers, who are you integrating to?

You can search CCO and this forum for some CUBE configuration examples and you will find plenty of similar posts.



New Member

SIP trunk

While CUBE will work, sometimes this is an overkill especially since you have to license it.  On the other hand, depending on what you are doing, CUBE could be required.  However, if you just need to place SIP calls from your CallManager, configuring your voice gateway as an IP2IP gateway will usually suffice. 

The reason you need CUBE or an IP2IP gateway is because your phones are talking H323 skinny and you are attempting to place the call via SIP.  The call will setup correctly as the phone places the call across the SIP trunk but then when the CallManager and gateway drop out to let the two end devices carry on the voice portion of the call, obviously it will fail since one end is talking SIP and the other is talking H323.  Therefore you configure your gateway to stay in the loop and do the conversion from SIP to H323 (and also possibly convert codecs also using transcoding).  This can be done easily via IP2IP.  For most of my customers, setting up CUBE would be a lot more involved and then there is also the licensing issue.  Here's a sample of the things that needed to be added for transcoding and IP2IP.  I believe this is fairly complete (to be added to an existing gateway which is already connected to a CallManager). This is what I use as my template for a new installation:

voice-card 0


dsp services dspfarm



voice call send-alert

voice rtp send-recv


voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip


  modem passthrough nse payload-type 98 codec g711ulaw


  bind control source-interface GigabitEthernet0/0

  bind media source-interface GigabitEthernet0/0


dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

maximum sessions 30

associate application SCCP


dial-peer voice 3 voip

translation-profile outgoing international

destination-pattern 9T

session protocol sipv2

session target

dtmf-relay rtp-nte

Hall of Fame Super Silver

SIP trunk

IP2IP GW configuration is CUBE, Cisco used to refer to it as IP to IP GW and since changed the terminology to Border Element (CUBE).  Either way it requires RTU licenses to be compliment (notice RTU is not enforced at this time).



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