cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
975
Views
0
Helpful
2
Replies

SIP / VoIP trunk > ISDN PRI via CME for TEHO aka Toll Bypass Help

DNAdatacom
Level 1
Level 1

hi all,

could really use some help and a sanity check here:

Calls come inbound to my CME via SIP dial peers over site to site VPN tunnel

Caller sends me 10 digits to my CME - matches outbound dial peer to my ISDN PRI

Call goes out and rings destination - ringback ok both ends of call

Call is answered and seems to connect fine - but no audio heard either direction

debugs and traces seem to insinuate a timer expires at approx 20 seconds into the connected call every time (call drops/disconnects)

But never audio present even after hold/resume - nothing

Can I assume SIP / dial peer configuration is correct if I get to this call connected stage?

I am forcing g711 with IOS dial peer command "codec g711" and I assume this should eliminate codec / transcoding issue ?

please advise

Sent from Cisco Technical Support iPad App

2 Replies 2

DNAdatacom
Level 1
Level 1

I should add that calls over same site to site VPN tunnel / SIP trunk to CME internal extensions work perfectly -

Audio is fine both ways

Only when call / pattern is directed out the PRI ISDN do I experience no audio

Sent from Cisco Technical Support iPad App

Hi Michelle,

What codec do you see on CME router for that call. Please check codec using "sh voice call summary" and see if codec is g711.The default codec used by the voip dial-peers is G.729 so you need to check call leg on voip side and force codec to negotiate G711.

It might be codec issue and you are not getting the RTP.

Regards,

Brajesh.