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Community Member

SIP endpoints & CUE == no DTMF/audio

Hi there,

I have a 2821 running CME 4.1 with CUE 3.0.1. I have a few SIP endpoints (Sipura & Linksys SPAs and a Mitel 5055) running off of it. Everything works perfect, except when any one of these endpoints tries to call voicemail (or any service DN connected to the Unity Express, for example the auto attendant). The SIP endpoint receives audio correctly, but is not transmitting any DTMF or voice to the CUE. Any ideas what might be the problem?

I have tested these scenarios (voice & DTMF):

SIP phone <-> SCCP phone (OK)

SIP phone <-> SCCP FXS port on local router (OK)

SIP phone -> PSTN via SIP trunk (OK)

SIP phone -> PSTN via FXO port (OK)

SIP phone -> CUE pilot number (no outbound audio, no DTMF)

SIP phone -> CUE auto attendant (no outbound audio, no DTMF)

<-> meaning that I have called to and from either endpoints.

Hall of Fame Super Red
Community Member

Re: SIP endpoints & CUE == no DTMF/audio

Thanks for the links, but my configuration is exactly as stated in those two links. Looks like the RTP is not reaching the CUE...

Community Member

Re: SIP endpoints & CUE == no DTMF/audio

Still not getting any luck. Any ideas? Anyone want to check out configs? If so, tell me what you need - I'll post it.

Community Member

Re: SIP endpoints & CUE == no DTMF/audio

Well, I fixed my DTMF issue. Played with the DTMF payload options and tested different modes of integration such as rtp-nte vs sip-notify. However, CUE is still not recieving any audio from my SIP phones...

Guess what - I just noticed that CUE isn't receiving any audio from any endpoint!!!

Anything I should look for/at?

Community Member

Re: SIP endpoints & CUE == no DTMF/audio

I have found the culprit!!!

In sip-ua, I had these lines:

nat symmetric check-media-src

nat symmetric role active

I negated these two commands and everything started working perfectly.

Needless to say, sip-ua options have a direct impact on CUE!...

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