Avaya SIP Interoperability

For this client our target architecture is based on Avaya Communication Manager in an ESS configuration acting as a single logical switch. We are using Genesys 7.6 components for IVR and in-queue treatments. The Genesys side is all SIP based e.g. GVP and Stream Manager. Effectively the Avaya and Genesys sides are seperate switches and we use external routing between them.

We have been having numerous problems with calls being dropped after 32 seconds – the standard SIP session timeout. We have fixed most of these problems but a SIP trace using wireshark (http://www.wireshark.org/) reveals that SIP interoperability is not 100% correct and the biggest problem is that SIP T-Server stays in the signalling path even when the call has been routed to a target back on the Avaya switch. This means that all calls in progress are dropped if we lose SIP T-Server e.g. a single point of failure.

We has been running CM5.1 with SIP Server 7.6 and the recommendation from Avaya and Genesys is to upgrade to CM5.2 and SIP Server 8 respectively. For information to upgrade to SIP Server 8 new license features are required including 8.0 ISCC and HA.

This week we thought that we would start by upgrading to CM5.2. Having done this we could not even get calls onto the SIP trunks – SES logs showed nothing. The upgrade process seemed to work OK and we checked and double checked all the settings. Still no joy.

Eventually we managed to get Avaya support involved and the problem seems to be related to SIP domain names. Basically, the domain name we had been using was not a valid fully qualified domain name (FQDN) e.g. ‘mysipdomain.com’. Whilst it was OK in CM5.1 it looks as though additional validation has been added in CM5.2.

If you are planning to upgrade to CM5.2 make sure you check you current SIP domain name is the places shows below!





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