Hi Wilson,From what I recall - Page 7

Anthony Brandelli · ‎09-18-2014

I've run into an issue where callers coming from the PSTN do not hear ringback after being transferred using a Cisco Unity call handler.

Ringback works internally between phones, and also works internally if I dial the call handler DN and request a transfer.

Call flow is this: SIP Provider -> CUBE -> SIP Trunk -> CUCM -> Unity Connection (SCCP)

The caller just hears dead air until the phone is picked up, or it bounces to voicemail.

There is an MRGL assigned to the SIP trunk to CUBE that contains an announciator and MoH.

The weird thing is the caller does hear about a second of MoH when Unity begins the transfer, but nothing after that.

debug ccsip messages:

1111111111 is the called party/number, 2222222222 is the calling number (my cellphone)

000925: Sep 18 14:08:06.696 MST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:1111111111@184.71.242.2:5060 SIP/2.0

Via: SIP/2.0/UDP 208.43.85.75:5060;branch=z9hG4bK27e0cc04;rport

From: "2222222222" <sip:2222222222@208.43.85.75>;tag=as2bec68a8

To: <sip:1111111111@184.71.242.2:5060>

Contact: <sip:2222222222@208.43.85.75>

Call-ID: 264119bc6be819f81634da4a67543569@208.43.85.75

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Remote-Party-ID: "2222222222" <sip:2222222222@208.43.85.75>;privacy=off;screen=no

Date: Thu, 18 Sep 2014 20:08:06 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Type: application/sdp

Content-Length: 361

v=0

o=root 1022 1022 IN IP4 208.43.85.75

s=session

c=IN IP4 208.43.85.75

b=CT:384

t=0 0

m=audio 11300 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

m=video 14308 RTP/AVP 34 99

a=rtpmap:34 H263/90000

a=rtpmap:99 H264/90000

a=sendrecv

000926: Sep 18 14:08:06.708 MST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:700@10.135.191.12:5061 SIP/2.0

Via: SIP/2.0/TLS 172.18.0.1:5061;branch=z9hG4bK40E577

Remote-Party-ID: "2222222222" <sip:2222222222@172.18.0.1>;party=calling;screen=no;privacy=off

From: "2222222222" <sip:2222222222@sip1.ixica.com>;tag=1D82524-20B8

To: <sip:700@10.135.191.12>

Date: Thu, 18 Sep 2014 20:08:06 GMT

Call-ID: 56A4B001-3EA611E4-8227D749-31CC0927@172.18.0.1

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 1453513609-1051070948-2183255881-0835455271

User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6a

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1411070886

Contact: <sip:2222222222@172.18.0.1:5061;transport=tls>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 69

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 394

v=0

o=CiscoSystemsSIP-GW-UserAgent 6324 6685 IN IP4 172.18.0.1

s=SIP Call

c=IN IP4 172.18.0.1

t=0 0

m=audio 16420 RTP/AVP 0 101

c=IN IP4 172.18.0.1

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

m=video 16422 RTP/AVP 34 119

c=IN IP4 172.18.0.1

a=rtpmap:34 H263/90000

a=fmtp:34 CIF=1

a=rtpmap:119 H264/90000

a=fmtp:119 profile-level-id=000000

000927: Sep 18 14:08:06.712 MST: //1040/56A2DB898221/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying

Via: SIP/2.0/UDP 208.43.85.75:5060;branch=z9hG4bK27e0cc04;rport

From: "2222222222" <sip:2222222222@208.43.85.75>;tag=as2bec68a8

To: <sip:1111111111@184.71.242.2:5060>

Date: Thu, 18 Sep 2014 20:08:06 GMT

Call-ID: 264119bc6be819f81634da4a67543569@208.43.85.75

CSeq: 102 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.2.4.M6a

Content-Length: 0

000928: Sep 18 14:08:06.716 MST: //1042/56A2DB898221/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying

Via: SIP/2.0/TLS 172.18.0.1:5061;branch=z9hG4bK40E577

From: "2222222222" <sip:2222222222@sip1.ixica.com>;tag=1D82524-20B8

To: <sip:700@10.135.191.12>

Date: Thu, 18 Sep 2014 20:08:06 GMT

Call-ID: 56A4B001-3EA611E4-8227D749-31CC0927@172.18.0.1

CSeq: 101 INVITE

Allow-Events: presence

Content-Length: 0

000929: Sep 18 14:08:06.728 MST: //1042/56A2DB898221/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 172.18.0.1:5061;branch=z9hG4bK40E577

From: "2222222222" <sip:2222222222@sip1.ixica.com>;tag=1D82524-20B8

To: <sip:700@10.135.191.12>;tag=6169~dc062259-9b0a-4df2-b05d-e4de329c1980-23733830

Date: Thu, 18 Sep 2014 20:08:06 GMT

Call-ID: 56A4B001-3EA611E4-8227D749-31CC0927@172.18.0.1

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

Allow-Events: presence

Server: Cisco-CUCM10.5

Supported: X-cisco-srtp-fallback

Supported: Geolocation

P-Asserted-Identity: "Cisco Unity" <sip:800@10.135.191.12>

Remote-Party-ID: "Cisco Unity" <sip:800@10.135.191.12>;party=called;screen=yes;privacy=off

Contact: <sip:700@10.135.191.12:5061;transport=tls>

Content-Length: 0

000930: Sep 18 14:08:06.732 MST: //1040/56A2DB898221/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 208.43.85.75:5060;branch=z9hG4bK27e0cc04;rport

From: "2222222222" <sip:2222222222@208.43.85.75>;tag=as2bec68a8

To: <sip:1111111111@184.71.242.2:5060>;tag=1D82538-D9C

Date: Thu, 18 Sep 2014 20:08:06 GMT

Call-ID: 264119bc6be819f81634da4a67543569@208.43.85.75

CSeq: 102 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: "Cisco Unity" <sip:800@184.71.242.2>;party=called;screen=yes;privacy=off

Contact: <sip:700@184.71.242.2:5060>

Server: Cisco-SIPGateway/IOS-15.2.4.M6a

Content-Length: 0

000931: Sep 18 14:08:06.804 MST: //1042/56A2DB898221/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK

Via: SIP/2.0/TLS 172.18.0.1:5061;branch=z9hG4bK40E577

From: "2222222222" <sip:2222222222@sip1.ixica.com>;tag=1D82524-20B8

To: <sip:700@10.135.191.12>;tag=6169~dc062259-9b0a-4df2-b05d-e4de329c1980-23733830

Date: Thu, 18 Sep 2014 20:08:06 GMT

Call-ID: 56A4B001-3EA611E4-8227D749-31CC0927@172.18.0.1

CSeq: 101 INVITE

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

Allow-Events: presence, kpml

Supported: replaces

Server: Cisco-CUCM10.5

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uas

Require: timer

P-Asserted-Identity: "Cisco Unity" <sip:800@10.135.191.12>

Remote-Party-ID: "Cisco Unity" <sip:800@10.135.191.12>;party=called;screen=yes;privacy=off

Contact: <sip:700@10.135.191.12:5061;transport=tls>;isFocus

Content-Type: application/sdp

Content-Length: 406

v=0

o=CiscoSystemsCCM-SIP 6169 1 IN IP4 10.135.191.12

s=SIP Call

c=IN IP4 10.135.191.12

b=TIAS:64000

b=CT:64

b=AS:64

t=0 0

m=audio 25280 RTP/AVP 0 101

a=ptime:20

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

m=video 0 RTP/AVP 31 34 96 97

a=rtpmap:31 H261/90000

a=rtpmap:34 H263/90000

a=rtpmap:96 H263-1998/90000

a=rtpmap:97 H264/90000

a=content:main

a=inactive

000932: Sep 18 14:08:06.808 MST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:700@10.135.191.12:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 172.18.0.1:5061;branch=z9hG4bK40F1BB8

From: "2222222222" <sip:2222222222@sip1.ixica.com>;tag=1D82524-20B8

To: <sip:700@10.135.191.12>;tag=6169~dc062259-9b0a-4df2-b05d-e4de329c1980-23733830

Date: Thu, 18 Sep 2014 20:08:06 GMT

Call-ID: 56A4B001-3EA611E4-8227D749-31CC0927@172.18.0.1

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

000933: Sep 18 14:08:06.812 MST: //1040/56A2DB898221/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK

Via: SIP/2.0/UDP 208.43.85.75:5060;branch=z9hG4bK27e0cc04;rport

From: "2222222222" <sip:2222222222@208.43.85.75>;tag=as2bec68a8

To: <sip:1111111111@184.71.242.2:5060>;tag=1D82538-D9C

Date: Thu, 18 Sep 2014 20:08:06 GMT

Call-ID: 264119bc6be819f81634da4a67543569@208.43.85.75

CSeq: 102 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: "Cisco Unity" <sip:800@184.71.242.2>;party=called;screen=yes;privacy=off

Contact: <sip:1111111111@184.71.242.2:5060>

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-15.2.4.M6a

Supported: timer

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 292

v=0

o=CiscoSystemsSIP-GW-UserAgent 4672 7011 IN IP4 184.71.242.2

s=SIP Call

c=IN IP4 184.71.242.2

t=0 0

m=audio 16416 RTP/AVP 0 101

c=IN IP4 184.71.242.2

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

m=video 0 RTP/AVP 34

c=IN IP4 184.71.242.2

000934: Sep 18 14:08:06.896 MST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received:

ACK sip:1111111111@184.71.242.2:5060 SIP/2.0

Via: SIP/2.0/UDP 208.43.85.75:5060;branch=z9hG4bK243fd8d0;rport

From: "2222222222" <sip:2222222222@208.43.85.75>;tag=as2bec68a8

To: <sip:1111111111@184.71.242.2:5060>;tag=1D82538-D9C

Contact: <sip:2222222222@208.43.85.75>

Call-ID: 264119bc6be819f81634da4a67543569@208.43.85.75

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Remote-Party-ID: "2222222222" <sip:2222222222@208.43.85.75>;privacy=off;screen=no

Content-Length: 0

000935: Sep 18 14:08:17.351 MST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: UPDATE sip:2222222222@172.18.0.1:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 10.135.191.12:5061;branch=z9hG4bK8347ed20ac

From: <sip:700@10.135.191.12>;tag=6169~dc062259-9b0a-4df2-b05d-e4de329c1980-23733830

To: "2222222222" <sip:2222222222@sip1.ixica.com>;tag=1D82524-20B8

Date: Thu, 18 Sep 2014 20:08:07 GMT

Call-ID: 56A4B001-3EA611E4-8227D749-31CC0927@172.18.0.1

User-Agent: Cisco-CUCM10.5

Max-Forwards: 70

Supported: timer,resource-priority,replaces

Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY

CSeq: 101 UPDATE

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED

Supported: X-cisco-srtp-fallback

Supported: Geolocation

Session-Expires: 1800;refresher=uac

Min-SE: 1800

P-Asserted-Identity: "Anthony" <sip:270@10.135.191.12>

Remote-Party-ID: "Anthony" <sip:270@10.135.191.12>;party=calling;screen=yes;privacy=off

Contact: <sip:700@10.135.191.12:5061;transport=tls>

Content-Length: 0

000936: Sep 18 14:08:17.351 MST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 10.135.191.12:5061;branch=z9hG4bK8347ed20ac

From: <sip:700@10.135.191.12>;tag=6169~dc062259-9b0a-4df2-b05d-e4de329c1980-23733830

To: "2222222222" <sip:2222222222@sip1.ixica.com>;tag=1D82524-20B8

Date: Thu, 18 Sep 2014 20:08:17 GMT

Call-ID: 56A4B001-3EA611E4-8227D749-31CC0927@172.18.0.1

Server: Cisco-SIPGateway/IOS-15.2.4.M6a

CSeq: 101 UPDATE

Allow-Events: telephone-event

Contact: <sip:270@172.18.0.1:5061;transport=tls>

Require: timer

Session-Expires: 1800;refresher=uac

Supported: timer

Content-Length: 0

000937: Sep 18 14:08:22.894 MST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: BYE sip:1111111111@184.71.242.2:5060 SIP/2.0

Via: SIP/2.0/UDP 208.43.85.75:5060;branch=z9hG4bK7d5b6278;rport

From: "2222222222" <sip:2222222222@208.43.85.75>;tag=as2bec68a8

To: <sip:1111111111@184.71.242.2:5060>;tag=1D82538-D9C

Call-ID: 264119bc6be819f81634da4a67543569@208.43.85.75

CSeq: 103 BYE

User-Agent: Asterisk PBX

Max-Forwards: 70

Remote-Party-ID: "2222222222" <sip:2222222222@208.43.85.75>;privacy=off;screen=no

Content-Length: 0

000938: Sep 18 14:08:22.898 MST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK

Via: SIP/2.0/UDP 208.43.85.75:5060;branch=z9hG4bK7d5b6278;rport

From: "2222222222" <sip:2222222222@208.43.85.75>;tag=as2bec68a8

To: <sip:1111111111@184.71.242.2:5060>;tag=1D82538-D9C

Date: Thu, 18 Sep 2014 20:08:22 GMT

Call-ID: 264119bc6be819f81634da4a67543569@208.43.85.75

Server: Cisco-SIPGateway/IOS-15.2.4.M6a

CSeq: 103 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=804,OS=128640,PR=693,OR=98712,PL=0,JI=0,LA=0,DU=16

Content-Length: 0

000939: Sep 18 14:08:22.898 MST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:700@10.135.191.12:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 172.18.0.1:5061;branch=z9hG4bK4102F8

From: "2222222222" <sip:2222222222@sip1.ixica.com>;tag=1D82524-20B8

To: <sip:700@10.135.191.12>;tag=6169~dc062259-9b0a-4df2-b05d-e4de329c1980-23733830

Date: Thu, 18 Sep 2014 20:08:17 GMT

Call-ID: 56A4B001-3EA611E4-8227D749-31CC0927@172.18.0.1

User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6a

Max-Forwards: 70

Timestamp: 1411070902

CSeq: 102 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=693,OS=98712,PR=804,OR=128640,PL=0,JI=0,LA=0,DU=16

Content-Length: 0

000940: Sep 18 14:08:22.902 MST: //1042/56A2DB898221/SIP/Msg/ccsipDisplayMsg: Received:

SIP/2.0 200 OK

Via: SIP/2.0/TLS 172.18.0.1:5061;branch=z9hG4bK4102F8

From: "2222222222" <sip:2222222222@sip1.ixica.com>;tag=1D82524-20B8

To: <sip:700@10.135.191.12>;tag=6169~dc062259-9b0a-4df2-b05d-e4de329c1980-23733830

Date: Thu, 18 Sep 2014 20:08:23 GMT

Call-ID: 56A4B001-3EA611E4-8227D749-31CC0927@172.18.0.1

Server: Cisco-CUCM10.5

CSeq: 102 BYE

Content-Length: 0

CUBE Dialpeers and voice config:

voice service voip
 ip address trusted list
 mode border-element 
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  registrar server expires max 1200 min 300
  rel1xx disable
  midcall-signaling passthru
  g729 annexb-all

dial-peer voice 1000 voip
 description *** Outbound CUCM ***
 destination-pattern ...
 session protocol sipv2
 session target ipv4:XXXXXX
 session transport tcp tls
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 2000 voip
 description *** Inbound CUCM ***
 session protocol sipv2
 session transport tcp tls
 incoming called-number ...
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 101 voip
 description *** Incoming Dial-Peer ***
 translation-profile incoming INCOMING1
 session protocol sipv2
 session target sip-server
 incoming called-number 1111111111
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 1 voip
 description *** Outbound Dial-Peer ***
 translation-profile outgoing Strip9
 destination-pattern 9.+
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!

Ayodeji Okanlawon · ‎10-25-2014

In addition to my email, I can see that unity connection is sending inactive SDP during the transfer, but CUCM doesn't send inactive SDP to CUBE. Lets work this out with TAC and hopefully we can resolve this without MTP

Please rate all useful posts

Wilson Gonzalez · ‎12-19-2014

Hi guys did you get any real resolution with TAC regarding the no ringback issue. I have a similar issue with CUC sending SDP with inactive back to an SME and the SME sends the SDP with inactive back to the originating cluster. Not sure what else to look for at this point but all my locales are setup to none. Just fyi.

Wilson

Anthony Brandelli · ‎12-26-2014

Hi Wilson,

From what I recall this was never directly addressed by TAC because it turned out there were other issues with both the ITSP and locales, I remember it being brought up on the TAC call but I don't think I ever heard an answer as to why this was happening/what this effected.

Thanks,

Anthony

Ayodeji Okanlawon · ‎10-24-2014

Did you do enable early offer support on your sip profile?

"Need early offer to check the above parameter : You may use "Early Offer support for voice and video calls (insert MTP if needed)" in the SIP Profile. "

Please rate all useful posts

Jonathan Pistagnesi · ‎07-18-2017

This is a late response but hopefully may help others looking for a possible solution...

In CUCM, look for service parameter "Duplex Streaming Enabled" - set to true

Jon

ksheth001 · ‎07-31-2017

Hi Jon,

I am going to try this tonight, we are facing this issue and hope this resolves it. Do I need to restart anything ?

Jonathan Pistagnesi · ‎07-31-2017

nothing to restart.

jon

ksheth001 · ‎07-31-2017

Jon,

I just tried this, I am not getting ringback unfortunately. I Dont think I am missing anything, is there any other step in between ?

Ayodeji Okanlawon · ‎08-01-2017

I suggest that you open a new thread. This an old thread and duplex streaming doesnt control ring back, please ignore that advice..

Open a new thread and explain your call flow, then describe the issues in detail. Does ring back fail on all calls? Does it fail on transfer?etc

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wilsongcisco · ‎08-29-2019

Hi All,

Something that I found that works is to change the Duplex Streaming Enabled to True under the CallManager Service Parameter, after this change ringback works when transfering from Unity

No Ringback from SIP Provider after Unity Transfer