Dial-peer assistance

[campbech1]

I'm very new to VoIP so go easy on me.

Here is our current dial-peer configuration. We are looking to just have any call from our SIP provider to be passed directly to CUCM. We are an organization with a lot of numbers all over several states. Maintaining a e164 list isn't a major issue but it will be quite a list and just the fear that a number is missed and then calls don't process.

Is there a better way to just pass through these incoming invites?

dial-peer voice 11 voip
 description Calls to CUCM (Pub)
 session protocol sipv2
 session target ipv4:10.19.254.10
 destination e164-pattern-map 200
 voice-class codec 1 offer-all
 voice-class sip asserted-id pai
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 12 voip
 description Calls to CUCM (Sub)
 preference 1
 session protocol sipv2
 session target ipv4:192.168.30.202
 destination e164-pattern-map 200
 voice-class codec 1 offer-all
 voice-class sip asserted-id pai
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 20 voip
 description Calls from NexVortex
 session protocol sipv2
 session target sip-server
 incoming called e164-pattern-map 200
 voice-class codec 1 offer-all
 voice-class sip asserted-id pai
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 31 voip

 description Calls to NexVortex

 translation-profile outgoing OUTGOING-SIP
 destination-pattern .T
 session protocol sipv2
 session target dns:nexvortex.com
 voice-class codec 1 offer-all
 voice-class sip asymmetric payload full
 voice-class sip asserted-id pai
 voice-class sip call-route url
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 no vad

[R0g22]

If you do not want a GW in between, terminate the SIP trunk directly to CUCM. AFAIK there is no way to bypass the INVITE's if a GW is in the path which is potentially acting as a b2bua.

[campbech1]

Maybe that wasn't phrased properly. I understand the GW will still have to process the call but is there a way to do this without maintaining every number on a e164 map that sits on the SIP trunk?

We will have several hundred numbers that sit on this SIP provider trunk. Since we have disaster recovery setup with the provider to send to three different GWs, in the event of a failure, we will need to maintain these on each router.

[Chris Deren]

Yes, if the CUBE/GW is only between CUCM and ITSP then your CUCM should be default-gw routing using e164 map or destination-target of +1......... (assuming you are on US dial plan and telco is sending +e164 strings or you've normalized on on the inbound dial-peer from telco) and your dial-peer to PSTN would match your country's routing with off-net access code used on CUCM, i.e. for US:

91[2-9]..[2-9]......

9011!

911

9911

etc. and you strip 9 via translation rule.  This way anytime you add new DID range or deploy new site there is nothing you need to add/change to this gateway.

[campbech1]

I understand, but not sure how to implement this based on our current dial-peers. Sorry, routing/switching/firewalls/wireless is what I typically do. VoIP is not in my wheelhouse at all.

Our current e164 maps just have the DIDs that we'd expect to receive from the SIP provider. This configuration might be totally wrong, but it's what we've come up with so far.

I've tried to modify the e164 map and still struggling.

I have the option of receiving the incoming call from our provider prepended with a 1 or without. Today we are translating this with them to drop the 1 and just send us the 10 digit string.

[Marcelo Morais]

Hi @campbech1,

 just an example using San Jose, CA Area Code (408 and 669) and a Phone with 10 digits number ...

On CUBE:

voice class e164-pattern-map 200
 e164 408.......$
 e164 669.......$

dial-peer voice 11 voip
 description Calls to CUCM (Pub)
 session protocol sipv2
 session target ipv4:10.19.254.10
 destination e164-pattern-map 200
 voice-class codec 1 offer-all
 voice-class sip asserted-id pai
 dtmf-relay rtp-nte
 no vad

On CUCM:

CUCM Administration > Device > Trunk > on the CUBE's SIP Trunk:
 Inbound Calls
  Significant Digits: 10
  Calling Search Space: <the CSS to access your Phones>

Hope this helps,
 Marcelo Morais