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Unity Express Problem

FXCL3725Smart
Level 1
Level 1

Topology:

SiteA:Call manager4.1.3<--------IPSec VPN---------->SiteB:Call Manager Express4.0(1)(on cisco1760 voice gateway)---Unity Express 2.1.3(on Cisco2650XM/w AIM-CUE).

Sympton:When the IP phones registered under Call Manager starts a call to IP phone regiestered under Call Manager Express, the call can be established but ,when the phone foward no-answer to voice mail , I cann't here the prompt of voice mail ; When we start a call from PSTN , the voice mail of the phone in CCME side can be reached and heard.

"voice service voip

allow-connections h323 to sip" command had been added in CCME router , but , It doesn't work.

All the codec are G.711u-law. Call Manager to CCME is H.323 trunk.

Can anyone told me why ? Urgent , thanks !!!

8 Replies 8

Erick Bergquist
Level 6
Level 6

Do you have the IP-to-IP gateway feature set? What it sounds like your doing is trying to pass a call from a VoIP call leg to a VoIP call leg on the router which the IP-IP gateway feature set is needed for to the best of my knowledge from above description.

Alternatively, if you had the CCM license for CUE then you could register CUE to CCM with JTAPI (CTI Route point) and get the calls from CCM to CUE that way, but in this case that won't really work since the CME phones work fine for VM except when the call comes from CCM.

If you have IP-IP feature set, does the router side show a call active between the 2 dial-peers (voip and sip). (show call active voice brief). does CUE have good route back to rest of network and can reach other IP phone?

CUE can reach the reset of the network.

I think there is only one h323 call leg and one Sip call leg.

btw, the VPN is established between 1760 and PIX.

Make sure the H323 interface binding on the gateway is setup to match the IP address of the H323 gateway in Callmanager.

Here is the command for binding.

interface GigabitEthernet0/0

h323-gateway voip bind srcaddr X.X.X.X

Regards,

Anup

I have binded the h323 address to lan ip address, otherwise the call between IP phones will not be connected.

My CME version is 4.0(1).Maybe the version is too higher?

Hello, im having the same problem. Did you find a solution to this issue. The only difference in my case is that the CME and the CUE are on the same router. CME 3.3, CUE 2.2.2 and CM 4.1(3), any suggestion

I just wanted to give this a little bump because we are having the same problem.

I setup a trunk between our call manager and a remote sites CME. I can call the phones just fine but when I call the voicemail number the call gets connected but I don't get any audio.

I have a transcoder in place to do g729 <-> g711 and I've enabled the voice service voip h323 to sip, h323 to h323, and sip to h323 but no luck!

Thanks!

did anyone manage to get a answer to this?

running 4.2(3) h323 trunk to ccme (4.2) with cue on the same box.

calls between the cluster and ccme worsk fine its just the forwarding to voicemail.

Calls to CUE only use G711 Codec, so you will need transcoding resources to support internetwork Calls

you can verify if your transcoder is working using the next command:

show dspfarm profile X

where the X is the profile configured for transcode

you will something like this:

Profile ID = 1, Service = TRANSCODING, Resource ID = 1

Profile Description :

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SCCP Status : ASSOCIATED

If this transcoder is working fine, probe adding a command in voice service voip

voice service voip

allow-connections h323 to sip

allow-connections sip to h323

supplementary-service h450.12

The las command is used for transfer other calls type

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