Hi to all,
we have just migrated our telephony lines from ISDN and h323 gateways to full SIP and now we have a following scenario:
PSTN --- SIP trunk --- CUBE ---SIP trunk --- CUCM
Almost everything works OK except calls from one provider who are also on SIP and in following scenario:
Incoming call from PSTN to our number which have a greeting message that is played outside CUCM, on IVR (UCCX). Caller hears that greeting, but when call needs to be connected to end user, both ends don't hear each other (call is established).
We investigated this situation and found that when message on IVR is ended, CUCM sends invite to CUBE and CUBE sends that invite to provider - so this should be invite for connecting both caller and called person. However, that invite from that caller does not have any SDP (both invites) in it so caller side responds with 100 OK but without SDP so there is no informations for connecting media.
With other providers we do not have any problems and we see that from our side there is same behaviour. After IVR message is played, we are sending invite from CUCM to CUBE and forward that from CUBE to PSTN without SDP. But, other providers responses with OK that contain SDP so media is connected and calling and called party can hear each other.
From our provider we have question if we can send SDP in invite to that problematic provider so they will then respond with OK and SDP. I'm not sure how could we do this?
Thanks!
Solved! Go to Solution.
When you transfer a call UCM will send a number of SIP messages to the CUBE to change the audio end point.
These messages only need to go to the CUBE as that is the device that switches the audio, they do not need to be sent on to the PSTN/SIP provider.
You can make the CUBE handle these messages without passing them on.
You need to make this change out of hours and test it thoroughly with inbound and outbound calls.
Add the following to the CUBE
voice service voip
sip
midcall-signaling block
If that causes problems, you can also try
voice service voip
sip
midcall-signaling passthru media-change
For details on Mid call signalling consumption see this link
Graham
When you transfer a call UCM will send a number of SIP messages to the CUBE to change the audio end point.
These messages only need to go to the CUBE as that is the device that switches the audio, they do not need to be sent on to the PSTN/SIP provider.
You can make the CUBE handle these messages without passing them on.
You need to make this change out of hours and test it thoroughly with inbound and outbound calls.
Add the following to the CUBE
voice service voip
sip
midcall-signaling block
If that causes problems, you can also try
voice service voip
sip
midcall-signaling passthru media-change
For details on Mid call signalling consumption see this link
Graham
Thanks,
with midcall-signaling block we resolved this situation.