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
____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.
__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.