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New Member

Calls to PSTN from QSIG Trunk Fails


We have integration between Cisco CUCM and Siemens PBX with Cisco 3845 ISR Router (also as SRST). The Gateway is Configured in MGCP. During normal operation, all the calling between Cisco IP Phones and Siemens extensions is fine. All the extensions from Cisco and Siemens are able to Dial to PSTN and receive calls.

The Cisco and Siemens talk to each other over QSIG. In SRST mode, Cisco IP Phones work fine, make calls to PSTN and receive calls from PSTN,able to call Siemens extensions (& vice-versa), Siemens extensions able to receive calls from PSTN.

But Siemens extensions are unable to dial to PSTN.

The Siemens and Cisco systems communicate using QSIG trunks configured on the Cisco 3845 router.

The error seen on the debug for the outgoing call originating from Siemens Extension is " **ERROR**: CCPCC_CallOrigination: SETUP timed-out (2nd T303) to NETWORK.  The SETUP failed."

The call flow is as follows:

Siemens extension ---> Cisco Voice Gateway (Port 0/2/0- Q-SIG Trunk) ----> Dial-Peer (destination pattern 9T and pointing to port 0/2/1) ---------> PSTN PRI E1 (connected to port 0/2/1). The calls fails with the above error.

Dial-peer voice 99999 pots

destination-pattern 9T

Port 0/2/1:!5

The calls from Siemens extension are routed over the QSIG trunk to E1 Port on the same router connected to ISDN PRI E1. The translations are working fine, the end digits sent to PSTN are the E.164 numbers.

Please help.


Re: Calls to PSTN from QSIG Trunk Fails

If it is only occurring when the phones are in SRST mode, there is probably an alias command in the SRST configuration that is hijacking the PSTN calls when the associated phone is registered via SRST.  Can you cut and past your configuration, or at least, the call-manager-fallback configuration?

Oh, wait, I just noticed that you are using MGCP normally.  So in SRST, it would fall back to H.323.  What dial-peer does the Siemens normally use via MGCP?

Cisco Employee

Re: Calls to PSTN from QSIG Trunk Fails

Hi there,

It seems that the q931 setup sent by GW on the E1-PRI is timing out due to

no response from the telco end.

Can u dial the same PSTN number from an IP phone and route it over that

E1-PRI and get it to work ?

If that's the case then perhaps we need to compare/check the SETUP msgs

being sent when Siemen's phone calls vs. CUCM IP phone.

debug isdn q931 on the GW for both scenario will help.

Pl. include GW config as well.


New Member

Calls to PSTN from QSIG Trunk Fails

The issue was due to the qsig messages not getting decoded. "qsig decode" command was included in the voice service voip. This resolved the issue.

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