I am configuring Mobile Voice Access for deployment on CUCM 8.6 and using a SIP Controlled ISR G2 Gateway with a PRI.
The cluster / Gateways has been in production for a while therefore have all the correct configuration to make calls and correct access-lists etc.
I have set-up MVA previously many times on a H323 Gateway with little or no problem but I am having trouble on a SIP Gateway.
Firstly is this (SIp Gateway) supported? Cisco documentation is very flakey.
The problem I am having is with making outbound calls, user authenticates fine and when dials number to call-out the call disconnects after around 5-10 seconds, this is for all types of calls (internal and PSTN)
- CUCM PUB is running MVA service and VXML Application on IOS points to PUB
- MVA Access number is 97144390877 (cannot add + as not allowed) MVA in Internal partition.
All DDI's are with + and Gateway sends + on all calls, everything inbound and outbound is expanded to E164.
- Service Parameters, Partial Match, 7 digits, same access number
- IOS, allow connections sip-sip, etc usual SIP voice service voip config that works for calls.
- Rerouting CSS is same as users regular CSS to make all / International calls
dial-peer voice 0877 pots
incoming called-number XXXX
dial-peer voice 0888 voip
session protocol sipv2
session target ipv4:CUCM SUB
This was pretty simple on H323 Gateway, the lack of documentation on SIP Gateways for MVA makes this more difficult, all help is greatly appreciated.
The MVA application is invoked and the call is sent to CUCM, therefore this lies with making the outbound call, what is checked to make this call? Is ishould be regular CSS and Route Patterns if I am correct.
If the CSS doesnt allow the call surely we would get the call can not be completed as dialled message.
My questions is what am I missing ? Is there something else I should be looking for? Any tips?
For SIP trunks, you must check the Redirecting Diversion Header Delivery - Inbound check box in the Trunk Configuration window.
The above link does not work
This is a SIP Controlled gateway, not a PSTN SIP Trunk, we are using a PRI for PSTN access.
Is this setting (Redirecting Diversion Header Delivery) for the SIP Trunk Gateway or only for SIP PSTN Trunks to allow for MVA to work?
Please follow the link for SIP gateway with MVA.
Hi, Please let me know the resolution.
Exact same scenario happening with me.
I am too using a SIP Gateway with a PSTN E1 PRI termation on it.
Call are getting dropped when post MVA IVR being played.
Any help would be appreciated.
I had this same exact problem. I was trying to follow guides I found online and all of them had me configure the same number for the MVA number in CUCM (in service parameters and in media resources-MVA) as well as in both dial peers.
What I was seeing was I could dial the MVA number from outside my system and I would get the IVR prompts. I could dial the outside number, but it would just connect be back to the beginning of the IVR again.
I ended up configuring a different unused DN in the media resources section for MVA. Once I did that, the outgoing call using the IVR worked.
Hope that helps anyone else with that issue.