Avaya ASM spits out "480 Temporarily Unavailable (SIPS not allowed)"

[learningSkype]

i'm in SIP hell...we converted several phones from H.323 to SIP protocol and the process was smooth. All the phones came up but now when i call the phones i get a busy signal. but the phones are in the same network region as my phone, and we're on the same CM, version 6.3.

ran some traces in ASM and i can see that in the INVITE the call shows "sips:14500@company.com" instead of sending just plain "sip:14500@company.com".

does anyone know why my call is appending an "s" on these SIP calls and sending "sips" when i'm just making an internal call, basically station to station?

 

[kyle555]

top right of the route pattern - enforce SIPS uri? is that set to Yes?

 

[learningSkype]

Secure SIP is set to no. but if i set it to Yes, what else am i impacting?

 

[learningSkype]

also, if you change the route pattern, don't you also have to make a change to the 46xxsettings file for Secure SIP (SIPS)?

 

[learningSkype]

i'm using 9611's in that office and some 9630 phones as well.

 

[kyle555]

Well, having SIPS as Y in the route forces CM to send out to the endpoint as SIPS and could be a problem.

Is it the same calling a SIP phone from a H323 or SIP? Is it the same calling a H323 phone from a H323 or SIP?

SIPS can be enforced on the phone end as much as the CM end, so it has to be skewed somewhere.

Are you requiring media encryption in your CM codec sets or in the SIP-phone-specific settings in 46xx? If so, and if you happen to be TCP from CM to SM, the TLS SIP phones to SM will never be able to make or receive SRTP calls to anything if they have to pump down a TCP link from SM to CM where the SRTP encryption keys are in the clear.

Just some ideas about where you might incompatible enforcement of security.