SIP Trunk with failover to PSTN Gateway Router as Second Choice

[blatkinson]

I have a need to send all calls that match a pattern first to a SIP trunk so they can be setup as video if the DN on the other side is for a codec. If the DN on the far side is not a codec or does not exist on the far side CUCM, I need to instead use my local Cisco H.323 PSTN GW configured in my local CUCM.

The problem is the audio-only call is rejected and my CUCM never attempts to use the PSTN GW when call routing is configured as outlined below. AAR is not an option under the phone config.

Send all calls to 9.1408XXXXXXX to the SIP trunk in case they are a video call. If 'far side' rejects the call because the dialed 408 DN is not a video unit or does not exist, I want to then be able to send the call to my PSTN gateway built in my CallManager.

The following two configurations complete calls if the call is a video unit dialing a 'far side' video unit, but does not complete if the call is a video unit performing an audio-only call to a 408 area code number:

1. Route List Option

_a. Route pattern 9.1408XXXXXXX points to a route list called B2B_RL.

_b. The B2B_RL contains two Route Groups in the following order, B2B_SIP_TRK_RG and OFFNET_PSTN_RG

_c. B2B_SIP_TRK_RG contains one member, B2B_SIP_TRK

_d. OFFNET_PSTN_RG contains one member, my PSTN gateway

_In summary:

__1. B2B_RL

___a. B2B_SIP_TRK_RG

____1. B2B_SIP_TRK

___b. OFFNET_PSTN_RG

____1. My PSTN gateway

2. Rerouting CSS Option - I have also tried creating a new CSS and new partition that contains a Route Pattern of 9.1408XXXXXXX that points directly to my PSTN gateway. In this case, I have the B2B Trunk setup with a Rerouting Calling Search Space of the new CSS. This also fails.

_In summary:

__1. B2B_SIP_TRK --> Rerouting CSS of 'Rerouting_B2B_CSS'

___a. Rerouting_B2B_CSS is the only CSS that can reach partition Rerouting_B2B_PT

____1. Rerouting_B2B_PT contains the Route Pattern 9.1408XXXXXXX which points directly to my PSTN gateway device.

I verified my PSTN GW setup by swaping the two Route Groups within the RL and audio calls complete and for sure utilize the PSTN GW (seen in debug isdn q.931)

I also verifed the same for the Rerouting_B2B_CSS by having the device temporarily use this new CSS.

DNA shows the call should complete to the SIP Trunk without issue, since it can't mimic the scenario.

[KonradStepniewski]

Video call goes to SIP trunk and this is working. Voice call goes to SIP trunk being rejected and ends - looks like this is correct too.

If you could return something else than reject then CM would reroute your call to next RG - PSTN.

I would look what SIP trunk returns to CM because your setup with RL and RG should works. I assume trunk returns call end/rejected and that's the problem.

[blatkinson]

Is there a way to rewrite their reject message(accomlished completely within CUCM... trying to avoid box-in-the-middle stuff), or instruct my CUCM to attempt further processing upon receiving reject? Maybe an enterprise parameter needs to be adjusted? far side is not willing to change reject message.

[KonradStepniewski]

Don't think it's possible with CUCM it self or at least I don't know any way to achieve this.

Solution could be CUBE doing this message rewrite but this is box-in-the-middle solution.