No ringback tone over SIP trunk

[bokoo]

Hello.

I have a DID/DOD connected to 3945 SIP(CUBE) and CUCM 8.0.

When I make a call from PSTN(Cell Phone) to our CUCM over SIP.

I don't hear any ringback, but the call is routed correctly and the call works fine once the called-party answers.

When I delete "progress_ind setup enable 3" at dial-peer 10, thne I can hear ringbacktone.

But that call was disconnect automatically after 20 sec while pickup that call from another phone in same pickup group.

RTP works fine during 20 sec before disconnect.

Attached file is a "debug ccsip mess" without "progress_ind alert enable 8"

I think CUBE receive BYE message from SP side with RESOURCE_UNAVAILABLE.

SP and we are just using G.711ulaw.

Any ideas are appreciated, thanks in advance!

--------------------------------------------------

voice service voip

allow-connections sip to sip

sip

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

dial-peer voice 2 voip

destination-pattern .T

session protocol sipv2

session target ipv4:SP SIP Server

session transport udp

voice-class codec 1 

voice-class sip asserted-id pai

voice-class sip early-offer forced

voice-class sip profiles 200

dtmf-relay rtp-nte

no vad

dial-peer voice 10 voip

destination-pattern XXXX

progress_ind setup enable 3

codec g711ulaw

session protocol sipv2

session target ipv4:CUCM

session transport udp

dtmf-relay rtp-nte

[David Chachanidze]

I have looked at the Debug Output and here is the problem:

Firts of all - "progress_ind setup enable 3" applies only to H323 Signaling, it has nothing to do with SIP.

After, I have noticed that the Provider is sending SIP Earler Offer (INVITE with SDP) and also is waiting for 180/183 with SDP as well to get Early Media, but the CUCM does not send the RINGING message with SDP, but sends SDP only after the call has been answered in OK message, which is retried several times (sent OK message multiple times towards ITSP), which ignores SDP in OK message as it was expecting SDP in RINGING/Progress Message.

That's why you have RTP until the OK message with SDP is retryed. But as soon as Provider times out for SDP to be received 180/183 Messages (which does not happen) it disconnects the call.

So here is the solution: Ask Provider to use the Delay offer, but I think the ITSP will not do that.

Hope you understand the problem, and also could find other solutions as well.

[acampbell]

Hi,

You could try setting the CUCM SIP REL 1XX Options

Sip profile --->

SIP Rel1XX Options

This field configures SIP Rel1XX, which determines whether all SIP provisional responses (other than 100 Trying messages) get sent reliably to the remote SIP endpoint. Valid values follow: •Disabled—Disables SIP Rel1XX. •Send PRACK if 1XX contains SDP—Acknowledges a 1XX message with PRACK, only if the 1XX message contains SDP. •Send PRACK for all 1XX messages—Acknowledges all1XX messages with PRACK. If you set the RSVP Over SIP field to E2E, you cannot choose Disabled.

Try setting to

Send PRACK if 1XX contains SDP

—Acknowledges a 1XX message with PRACK, only if the 1XX message contains SDP.

Regards

Alex