I have an Xmedius Fax server that is connected with a SIP trunk to the CUCM cluster.
I am able to receive faxes no problem from the PSTN, but when I place a call out XMEDIUS-SIP-CUCM-MGCP-PRI I get a "temporary failure" message in the Q931 debug.
All other PSTN calls work fine.
Calling number is sent as 10 digits and called party is 11 digits, as all other calls the system. Any ideas?
Sep 15 14:31:47.700: ISDN Se0/1/1:23 Q931: TX -> SETUP pd = 8 callref = 0x11BD
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98396
Exclusive, Channel 22
Calling Party Number i = 0x2883, '631XXXXXXX'
Called Party Number i = 0xA8, '1631XXXXXXX'
Sep 15 14:31:49.071: ISDN Se0/1/1:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x11BD
Cause i = 0x80A9 - Temporary failure
Sep 15 14:31:49.097: ISDN Se0/1/1:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x11BD
Are your Route Patterns dropping any digits? They are usually dropping the initial Trunk Access code "9" or "0". Take a look at the Route Pattern the call matches and see where the "." is and if it is dropping pre-dot.
Also do you know what the Calling Party is supposed to look like? Most PSTN's need a specific amount of digits to identify you as a customer. This can sometimes be as small as the last 4 digits. Anything longer or shorter the PSTN PBX sometimes drops the call. You can check how many they expect by running the same debug on an incoming call, the Called Party on an incoming call will contain the number of digits being sent from the far side PBX. You will then need to adjust Call Manager respectively for outbound Calling Party transformation mask.
Hope this helps.
After working with Cisco, it appears to be an issue related to the configuration being setup as so:
XMEDIUS - SIP - CUCM - Q931 - PRI - PSTN
Fax server doesn’t support SIP UPDATE messages seems to be, and sends that 481 Error code back to CUCM which, forces it to disconnect the call.
They currently tell CUCM they support it, however when CUCM sends it the Xmedius rejects it and forces the CUCM to disconnect the call.
This cannot be changed on the CUCM as it’s a reaction to the PROGRESS message received from the PSTN. The Ladder Diagram above shows that exchange.
I am working with Xmedius to resolve this. A work around I guess would be to use H.323 between the fax server and the CUCM.
Hi, thanks for posting what you ran into with Xmedius and CUCM. We just ran into this issue as well.
Here's an update to this since it's always infuriating to come to the support forums with a problem and never see anyone come back and post a proper solution.
Add this registry key into the XMedius server as follows:
(this makes Xmedius ignore the SIP UPDATE message)
REG_SZ (String value)
The list of SIP requests to use in the "Allow" header of SIP messages sent by the FoIP Driver.
(Default: INVITE, ACK, BYE, CANCEL
Thanks very much to both darren360 and jcp408ADP for this solution.
Worked perfectly for a customer of mine, and saved me an evening of troubleshooting.