Dial out without ARS access code

[ejvl]

Hi,
We’ve a problem in our configuration, Avaya in VMWARE environment, Software Load: R016x.03.0.141.0 with Session manager.
The ARS access code is “0”. So if we want to dial to 0612345678, we’ve to dial 00612345678.
With our 9608 H323 phones, it works fine, but we’ve also some 9608 phones with SIP firmware.

When the H323 phones want to dial out without the ARS access code, we’re receiving a busy tone, that works fine.
But when our SIP phones want to dial out without the ARS access code, the call is routing to the trunk, and users can call without ARS access code.

Here’s a trace, we're call from station 9685 (9608 with SIP Firmware) to 0612345678, without ARS access code.
We've received an denial event 1751, no AAR/ARS route, that's true because 06xxxxxxxx is not in the ARS, but then the term trunk-group, so the call is routing to the trunk without the ARS access code.

14:13:43 TRACE STARTED 06/13/2018 CM Release String cold-03.0.141.0-22506
14:13:44 SIP<INVITE sip:9685@sip.xxx.com:5061;avaya-cm-fnu=off-hook S
14:13:44 SIP<IP/2.0
14:13:44 Call-ID: 96_5b21269771fc47b5303y5k346bp5656484q6o304019
14:13:44 252e1w5d_I968510.6.5.188
14:13:44 SIP>SIP/2.0 183 Session Progress
14:13:44 Call-ID: 96_5b21269771fc47b5303y5k346bp5656484q6o304019
14:13:44 252e1w5d_I968510.6.5.188
14:13:44 SIP>PUBLISH sips:9685@sip.xxx.com SIP/2.0
14:13:44 Call-ID: 05c50224b79e8142f565dab3100
14:13:44 active station 9685 cid 0x1b0
14:13:44 SIP<SIP/2.0 200 OK
14:13:44 Call-ID: 05c50224b79e8142f565dab3100
14:13:50 SIP>SIP/2.0 484 Address Incomplete
14:13:50 Call-ID: 96_5b21269771fc47b5303y5k346bp5656484q6o304019
14:13:50 252e1w5d_I968510.6.5.188
14:13:50 SIP<INVITE sip:0612345678@sip.xxx.com:5061 SIP/2.0
14:13:50 Call-ID: 96_5b21269771fc47b5303y5k346bp5656484q6o304019
14:13:50 252e1w5d_I968510.6.5.188
14:13:50 SIP>SIP/2.0 100 Trying
14:13:50 Call-ID: 96_5b21269771fc47b5303y5k346bp5656484q6o304019
14:13:50 252e1w5d_I968510.6.5.188
14:13:50 dial 06 route:ARS
14:13:50 denial event 1751: No AAR/ARS route pat/pref D1=0x9f3a D2=0x6a000000
14:13:50 term trunk-group 1 cid 0x1b0
14:13:50 seize trunk-group 1 member 53 cid 0x1b0
14:13:50 Calling Number & Name NO-CPNumber NO-CPName
14:13:50 SIP>INVITE sip:0612345678@sip.xxx.com SIP/2.0


Any idea how this is possible?

Thanks!

 

[kyle555]

If you 'display locations', what is the proxy sel rte? That's the default where CM goes in the absence of any other stuff being defined and could explain why you're going there.

Consider: a SIP phone has to get a dialplan to send requests. H323 phones get each digit interpreted by the PBX, so you dial 0 (PBX decides that's ARS) then any other digits pass along to the ARS table. With a SIP phone, it frames a complete invite with all the digits 012345678. CM might have an ARS match for 012345678, but not 12345678, and so your SIP phone never matches something in ARS and tries the 'last way out' as the proxy selection route pattern.

 

[ejvl]

Thanks for your response.
The proxy sel rte in the “display locations” is empty, nothing configured. In the trace we can see the call is routing to trunk 1, and trunk 1 is the Session Manager in our configuration.
We’ve make a test, and put there a route pattern to another trunk (PSTN) and make a trace again, but again the call is routing to trunk 1, so it seems like, there’s no difference in routing….