i have ccm 220.127.116.110-13 and have mobile connect running like a champ. i need mobile access to toggle the mobile connect on and off. issue is the call comes in and the vxml kicks off and then...the call drops.
debug call app voice vxml sees the call get to the gw (2811 running 12.4.15T5), kick off the app and attempt to play the audio files, but no luck.
As you stated, the documentation doesn't really tell you what the DN/DID settings for Mobile Voice Access should be. The only way I could make this work was to translate the dialable DN for Mobile Voice Access into a dummy extension for use only by the VXML application. I put the dummy extension in two places: the Mobile Voice Access Directory Number (in Media Resources) and the Mobile Voice Access Number in Service Parameters//CallManager Service/Clusterwide Parameters (System - Mobility).
I added a route pattern for the Mobile Voice Access DID in a partition that is dialable internally and externally, just like a phone with a DID. Set the called party transform mask to the DN you set in Media Resources and Service Parameters. Since I didn't configure Application Dial Rules, I put a 9 in the prefix digits line of the calling party transformations. This works with cell phones inbound since the remote destinations are all prefixed with a 9. To enable automatic identification for calls from a person's internal desk phone, add a translation profile like the one below to convert the user's extension to one of their configured remote destination numbers. In the example, extension 3577 is associated to remote destination number 202-456-1111. Notice that the 9 is included because the route pattern always prefixes it. I don't think wildcards are worth the trouble here because it has to be a 1-to-1 mapping anyway.
The H.323 gateway config should have an inbound calling search space that includes the MVA number but it does not need to include outside numbers. Apparently that is handled by the MVA extension once the call comes back from the H.323 gateway.
If you set "Matching Caller ID with Remote Destination" in Service Parameters to Partial Match, the cell phone is forked onto the mobility-enabled phone very early in the process, like right after the call to MVA completes, even before signing in with your PIN. When this happens, calls placed via MVA from outside create two call appearances on the mobility-enabled phone, one normal and one remote linked call. The second one looks sort of like a shared line where the available softkeys are Barge and New Call, but you can't use them. Strange! It works fine when set to Complete Match, which is the default.
Here are the commands I added to my working MGCP gateway:
In the returned file you should see various URL's that refer to audio files.
I found that these had "null" as the localisation.
I set the localisation under Media Resources/Mobile Voice Access and in the vxml gateway configuration (not sure which one fixed this) on Call Manager. I then had to reboot the Publisher (maybe just restarting Cisco Unified Mobile Voice Access Service would have been sufficient) and I restarted the gateway to be sure.
You should now see the correct localisation appearing in the audio file URL's in the vxml file mentioned above. You should now be able to browse to these audio files and listen to them.
Hi, I also have the connect up and running with no problems. I have also gone through all of the steps listed in the CUCM Features manual. I also had the null problem when just german was selected, however when I turned on English the null was replaced with eng.
However, my call are not making it to the service. I am not seeing the counter raise at the sho call applic voice cmm command. Did you guys have to load something on the router or was the url to the CUCM enough. I keep getting a message that the participant is not reachable/connected (out of order).
Any clue what I am missing?
All of the CUCM side settings are done and below are the relavant portions of my config:
IntroductionCUCM Routing RulesDial String implementation PolicyCUCM Routing LogicSIP URI Call Routing Analysis+++ Case Study: 1 ++++++ Case Study: 2 +++Conclusion
Over the last few months, I have had the privilege of working on SI...
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