Static SIP connecton without authentication

[CyberNBD]

Hi all,

my providers supports static SIP connections based on IP, without the need for credentials / authentication.  Is there a way to configure this on a cisco router?

I now have multiple voip lines / trunks with matching credentials for incoming calls but I can only supply only one authentication for outgoïng calls since all lines and trunks use the same realm.

Skipping the whole sip-ua registration process would help me to be able to make outbound calls on all numbers instead of beïng stuck to one line or trunk for outbound calls. 

Any other suggestions on how to enable multiple authentications with the same realm are also welcome (I tried authentication on dial-peer-level but that doesn't seem to work).

Thanks,

Tom

[R0g22 · Accepted solution]

Use SIP multi tenant -
https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-multi-tenants.html

You might need to upgrade your IOS in order to leverage the feature.

[CyberNBD]

Thanks,

The SIP multi tenant configuration seems to work, however outgoing calls are now quite randomly distributed through the SIP peers. 

I now have configured two voice class tenants: tenant 1 for SIP account X and tenant 2 for SIP account Y. 
I also configured two dial-peers, again I linked dial-peer 1 to tenant 1 and thus SIP account X and dial-peer 2 to tenant 2 and thus SIP account Y.

Is there an easy way to configure that when for example extension 1195 makes an outbound call it has to go though sip account Y (dial peer 2) and all the rest goes throug dial-peer voice 1?

It seems it is perfectly possible to do outbound dial-peer selection based on destination, but I need to do dial-peer selection based on source.

[R0g22 · Accepted solution]

Use incoming URI to match specific incoming dial-peers that further use a DPG config to select outgoing dial-peers.

[CyberNBD]

I tried two approaches now:

1. adding answer address to outgoing dial-peer.  Result: dial-peer matches perfectly based on calling number (ANI) but then switches randomly to outgoing dial-peer. This was before your DPG solution.

Config:

dial-peer voice 101 voip
 description VOYS CYBER OUTGOING
 answer-address 003185401166.
 destination-pattern .T
 session protocol sipv2
 session target dns:sipproxy.voipgrid.nl
 session transport udp
 voice-class codec 1  
 voice-class sip tenant 1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 102 voip
 description VOYS AU OUTGOING
 answer-address 0031302270195
 destination-pattern .T
 session protocol sipv2
 session target dns:sipproxy.voipgrid.nl
 session transport udp
 voice-class codec 1  
 voice-class sip tenant 2
 dtmf-relay rtp-nte
 no vad

So I tried the DPG, but now outgoing calls aren't processed anymore (busy tone).  Only when I add the destination-pattern .T to dial peers 1 and 2 (which shouldn't be necessary since these are dial-peers for incoming selection) calls are processed but again perfectly matched incoming to random outgoing.

voice class dpg 101
 description CYBER OUTGOING DIAL PEER GROUP
 dial-peer 101
!
voice class dpg 102
 description AU OUTGOING DIAL PEER GROUP
 dial-peer 102
!
dial-peer voice 101 voip
 description VOYS CYBER OUTGOING
 destination-pattern .T
 session protocol sipv2
 session target dns:sipproxy.voipgrid.nl
 session transport udp
 voice-class codec 1  
 voice-class sip tenant 1
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 102 voip
 description VOYS AU OUTGOING
 destination-pattern .T
 session protocol sipv2
 session target dns:sipproxy.voipgrid.nl
 session transport udp
 voice-class codec 1  
 voice-class sip tenant 2
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 1 voip
 description MATCH INCOMING CYBER ANI TO OUTGOING DIAL PEER
 answer-address 003185401166.
 destination dpg 101
!
dial-peer voice 2 voip
 description MATCH INCOMING AU ANI TO OUTGOING DIAL PEER
 answer-address 0031302270195
 destination dpg 102

what am I missing here?

[Carlo Poggiarelli]

Hi,

On incoming dialpeer you can apply a translation profile that add a prefix to called number.

Than you can configure an outgoing dialpeer matching the added prefix and remove it applying another translation profile before the invite to the TSP.

HTH

Regards

Carlo

[CyberNBD]

Got the DPG config working

Needed to add

voice-class codec 1 

to the incoming dial-peers 1 and 2