sip trunk between CUCM 7 and CME (with VG224)

[Sergey Balyura]

Hello, all!

I need to maintain a sip trunk between CUCM 7.1 and CME with vg224.

I've done sip trunk on CUCM, 've created route pattern.

When I try to call from CUCM to analog phone on vg224 this phone is ringing.

but the call is terminated immediately when I get an analog phone off hook.

I run deb ccsip verbose.

here is my logs - first one is when the analog phone is ringing, the second one - I picked up the phone.

and on CME I have the next dial-peer

dial-peer voice 700 voip

incoming called-number 700...

codec g722-64

please help me to understand what I'm doing wrong way.

[Gajanan Pande]

Sergey,

Call drop as soon as the called phone goes off-hook means Codec Mismatch between calling & called party. Pls check codec config. You will have to use a common codec or a Transcoder if both phones cant use same codec.

GP.

Pls rate the post if it helps !!

[Sergey Balyura]

thank you! I solve the issue with the immediate disconnection.

but I receive two more issues.

1. after the call is set up both sides cannot hear each other.

2. when I put the handset on the hook my VG224 line is stuck in "EFXS_WAIT_RELEASE_REQ" state...

I can imagine thos is about connectivity between endpoints... but ping is working correctly.

please help me