Incoming call PSTN transfered to branch office via AA (unity connection) are dropped

[filiberto.aguirre]

Hi,

End user has CUCM 11.5, Unity connection 11.5, and voice gateway 2921; all in  main office. There is a branch office with voice GW 4351 config as SRST that communicate with main office  via VPN. All the ip phones are 78xx. In both voice GW there is a SIP trunk with PSTN.

- calls between IP extension main office and branch office complete well

- calls from PSTN (sip trunk) to main office complete well.

- calls from PSTN (sip trunk) to branch office complete well.

- calls from PSTN (sip trunk) to IP extension on main office via AA complete well.

The issue:

when an incoming call from PSTN on main office and goes to menu of AA (unity connection) and dials an extension number that correspond to branch office the call is dropped.

We have tried with differentes codecs but not success

Any suggestion?

regards

[Dennis Mink]

Can you add the debug ccsip info of a failed call and provide dialing and dialed number.

cheers

[filiberto.aguirre]

Hi Dennis,

The issue was solved,  but now we detected the following issue.

When the incoming call from PSTN (sip trunk)  is received in branch office  then redirected to AA (unity connection on main office)  an then via the menu of AA  the ip extension of branch office is dialed

when the called party answer  nobody can hear each other.

For the moment  we have tried changing codec on regions but not succes up to now.

[filiberto.aguirre]

Hi Dennis,

I attach the some screenshoot for failed a success call and the sh run of the GW.

we have tried with codec on dial-peer an region . the trace between success an failed call are very similar. The difference is:

- incoming call from PSTN  on Main Office --> AA (unity main office) --->dialed by menu of AA ip ext of branch --> call Ok.  The trace shows calling number and called number.

-- incoming call from PSTN  on branch Office --> AA (unity main office) --->dialed by menu of AA ip ext of branch --> call FAILED. Trace shows  just calling number.

any suggestion?

[Chris Deren]

Is the Unity Connection SIP trunk (assuming SIP integration) or SCCP voicemail port (if using SCCP) configured with correct CSS with access to the phones' partition? How about Unity restriction tables, are they defined to allow these calls? Are the entered extension matching Unity Connection voicemail user extensions, if not is the Call Handler setup to allow transfers to non-user extensions?

[filiberto.aguirre · Accepted solution]

The problem was cleared after disabling  SIP inspection in both FW (fortinet) that handle the VPN between  main site and branch.

regards