No ringing back tone

[Alexander Afanasenkov]

Hello,

I'm having a problem in my topology;

IP Phone -> CUCM -> CUBE -> SIP Trunk -> Rostelecom

CUCM 11.0.1
CUBE (CISCO 2901) Version 15.4(3)M4

I am getting no ring back tone when I make calls between ip phones or pstn.

I need your experiences. Can anyone help me to solve it?
Thanks in Advance.

[Rajan]

Hi Alexander,

I checked the debugs. We are indeed getting SDP in the 183 message from provider and then I could see this:

025673: Oct 27 08:16:28.643 MSD: //17692/DCA009800000/CCAPI/ccCallCutProgress: Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0 Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)

Can you try the things mentioned in this discussion. 

https://supportforums.cisco.com/discussion/11528476/outbound-ringback-issues-over-sip-trunk-using-cucm-86-and-h323-gateway

Disable early media and try invoking annunciator in CUCM.

disable-early-media 180

HTH

Rajan

Please rate useful posts

[Alexander Afanasenkov]

Hi Rajan,

Should I remove 'early-offer forced'?

voice service voip
 sip
  early-offer forced

I also added 'disable-early-media 180' into sip-ua section.

No luck.

[Rajan]

Hi Alexander,

Not needed. have you also checked the annunciator part in CUCM ?

[Alexander Afanasenkov]

Hi Rajan,

Yes, trunk already have MRGL with: 

ANN_2

CBF_2

IVR_2

MOH_2

MTP_2

tver-br1-xcode

tver-br1-xconf

tver-br1-xmtp

sh sccp 

SCCP Admin State: UP

Gateway Local Interface: Loopback11

        IPv4 Address: 10.8.127.8

        Port Number: 2000

IP Precedence: 5

User Masked Codec list: None

Call Manager: 10.24.90.2, Port Number: 2000

Priority: N/A, Version: 7.0, Identifier: 2

Trustpoint: N/A

Call Manager: 10.24.90.1, Port Number: 2000

Priority: 1, Version: 7.0, Identifier: 1

Transcoding Oper State: ACTIVE - Cause Code: NONE

Active Call Manager: 10.24.90.1, Port Number: 2000

TCP Link Status: CONNECTED, Profile Identifier: 1

Reported Max Streams: 6, Reported Max OOS Streams: 0

Supported Codec: g729abr8, Maximum Packetization Period: 60

Supported Codec: g729ar8, Maximum Packetization Period: 60

Supported Codec: g711alaw, Maximum Packetization Period: 30

Supported Codec: g711ulaw, Maximum Packetization Period: 30

Supported Codec: g729br8, Maximum Packetization Period: 60

Supported Codec: g729r8, Maximum Packetization Period: 60

Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30

Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30

Conferencing Oper State: ACTIVE - Cause Code: NONE

Active Call Manager: 10.24.90.1, Port Number: 2000

TCP Link Status: CONNECTED, Profile Identifier: 2

Reported Max Streams: 8, Reported Max OOS Streams: 0

Supported Codec: g729br8, Maximum Packetization Period: 60

Supported Codec: g729r8, Maximum Packetization Period: 60

Supported Codec: g729abr8, Maximum Packetization Period: 60

Supported Codec: g729ar8, Maximum Packetization Period: 60

Supported Codec: g711alaw, Maximum Packetization Period: 30

Supported Codec: g711ulaw, Maximum Packetization Period: 30

Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30

Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30

TLS : ENABLED

MTP Oper State: ACTIVE - Cause Code: NONE

Active Call Manager: 10.24.90.1, Port Number: 2000

TCP Link Status: CONNECTED, Profile Identifier: 3

Reported Max Streams: 34, Reported Max OOS Streams: 0

Supported Codec: g711ulaw, Maximum Packetization Period: 30

Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30

Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30

Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30

TLS : ENABLED

[Rajan]

Hi Alexander,

Can you check whether the option "Disable Early Media on 180" is checked on the SIP profile configuration which is used for that SIP trunk and let us know.

[Alexander Afanasenkov]

Hi Rajan,

I confirm option "Disable Early Media on 180" is checked, also i include screenshots SIP Trunk/Profile in my first post.

Thank you.

[Ayodeji Okanlawon · Accepted solution]

Alex,

This is not a configuration issue. This is how ringback works over sip trunks..

When CUCM receives 180 ringing (without SDP) CUCM plays rigback locally. It instructs the ip phone to play ring back..

When CUCM receives 183 with SDP, it tells the phone to cut through audio and listen to whatever the far end is sending. If the far end is sending nothing in 183 with SDP, then you hear nothing.. If your ITSP is sending 183 with SDP, then CUCM is right to listen to whatever the other end is sending in the SDP. 

So there is nothing you are going to do to change this other than to get your ITSP to send 180 ringing without any SDP or for then to actually play ring back in their 183 with SDP.

The disable early-media 180 can only affect 180 message with SDP. So in this case it is not relevant

[Alexander Afanasenkov]

Thank you for your replay, does it mean, that i must contact with ITSP and ask them to make the necessary changes?

[Ayodeji Okanlawon]

Yes please do. Tell them they either need to send 180 ringing or send ringback in 183 with sdp

[Alexander Afanasenkov]

I take a wait for a response from the ITSP.

Thank you.

[Alexander Afanasenkov]

Hello,

ITSP replied that at the moment there is a problem with ringback tone on his side.

Thank you all for your help.

[Suresh Subramanian]

in the SIP Profile assigned to the CUBE SIP Trunk can you set the parameter 'Early Offer support for voice and video calls' to "Mandatory (insert MTP if needed)" and check? Also keep the command 'early-offer forced' under SIP, don't remove it.

[Alexander Afanasenkov]

Hi Suresh,

Added "Mandatory (insert MTP if needed)", SIP Profile and Trunk i reset and restarted.

Problem still there.

Thank you.

[Suresh Subramanian]

I think it could be because of PRACK enabled on the SIP profile. could you please try setting the option "SIP Rel1XX Options" to 'Disabled' and try again?

EDIT: could you please attach the CUBE configuration as well?