No Ringback from SIP Provider after Unity Transfer - Page 4

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
!

Anthony Brandelli · ‎10-17-2014

To me this is service impacting as well - I have been trying to be patient with TAC especially since I suspected this may have been a bug and I understand those aren't fixed instantly.

My Cisco partner manages the interaction with TAC as part of a contract I have with them, but the delay is not coming from them (my Cisco partner), I can tell this for sure based on the email timestamps in the conversation between them and TAC... (And me)

I don't mean to point fingers here, just posting this for full transparency. I've poked them again for another update, won't hold my breath though because it's Friday. If poking doesn't work maybe I'll start throwing office supplies... (Kidding)

Anthony

Suresh Subramanian · ‎10-17-2014

Good luck Anthony, eagerly waiting fir the solution. Please let us know. :-)

//Suresh Please rate all the useful posts.

Ayodeji Okanlawon · ‎10-17-2014

Anthony,

If you do not get any feedback by Monday, let me know and I will open a case with cisco. I will need your email to send the webex invite..You can PM me. hopefully we can resolve this by monday

Please rate all useful posts

Anthony Brandelli · ‎10-17-2014

I'd be happy to take you up on that - I'll keep everyone posted on this.

I'll shoot you a PM on Monday with my email depending on how this plays out here.

Anthony

Manish Prasad · ‎10-18-2014

Anthony can you give one more try with below changes and let me know.

voice class sip-profiles 5
response 200 sdp-header Audio-Attribute modify "inactive" "sendrecv"

dial-peer voice 1000 voip

voice-class sip profiles 5

And attach these outputs here.

1.debug ccsip message

2."show media streams" and "utils network capture eth0" through SSH login to CUCM server where ANN is registered and when call is transferred from call handler.

Anthony Brandelli · ‎10-18-2014

I added the requested config to CUBE and unfortunately there is still no ringback present.

Find attached the requested debug information. Time of call 3:07PM, other information remains the same as previous from thread.

I'm somewhat puzzled because "show media streams" seems to indicate an active ANN stream...? Or am I reading this wrong?

Thank you,

Anthony

Manish Prasad · ‎10-19-2014

Yes there is an active ANN stream from 10.135.191.12:24580 to 172.18.0.1:18690 and which also can be verified by checking the RTP stream from the packet capture after the exchange of SIP ACK message.

Now what puzzling me here is why ITSP sends media=inactive in response to CUCM DO Re-Invite. Now that something you can check with Telco but just for a workaround can you do one more change and let me know.

voice class sip-profiles 6
request REINVITE sdp-header Audio-Attribute modify "inactive" "sendrecv"

dial-peer voice 1 voip (if this is your outgoing dial-peer towards ITSP)
voice-class sip profiles 6

Now attach ccsip message only.

Anthony Brandelli · ‎10-19-2014

As requested I added the config to the dial peer facing the ITSP. There is still no ringback present but attached is the ccsip messages output.

Thanks,

Anthony

Manish Prasad · ‎10-19-2014

Because sip profile 6 did not kicked in to modify DO Re-Invite.

Did you applied it to correct dial-peer ?( You can check with debug voice ccapi inout). If dial-peer is correct then modify sip-profile 6 with this...

request any sdp-header audio-attribute modify "inactive" "sendrecv"

response any sdp-header audio-attribute modify "inactive" "sendrecv"

My purpose in modifying those parameters is to extend media stream till ITSP interface , which remain terminated at CUBE inside interface.

Do attach ccsip message.

Anthony Brandelli · ‎10-19-2014

dial-peer voice 1 voip
description *** Outbound Dial-Peer ***
preference 10
destination-pattern .+
session protocol sipv2
session target sip-server
voice-class sip profiles 6
dtmf-relay rtp-nte
codec g711ulaw
no vad

It looks like it is under the right dial peer so I modified as suggested again. (Unless I have misunderstood, this should be the outgoing dial peer towards my provider, not the incoming dial peer from the provider, and not the dial peer facing CUCM, apologies if I'm off the mark)

Class 6 now looks like:

voice class sip-profiles 6
request ANY sdp-header Audio-Attribute modify "inactive" "sendrecv"
response ANY sdp-header Audio-Attribute modify "inactive" "sendrecv"

There is still no ringback present (assuming I have this configured right).

Attached are ccsip messages output from a call at 10:36PM.

Thanks,

Anthony

Manish Prasad · ‎10-19-2014

Anthony can you give one more last try from my side. Add these in sip profile 6.

voice class sip-profiles 6
request ANY sdp-header Audio-Attribute modify "inactive" "sendrecv"
response ANY sdp-header Audio-Attribute modify "inactive" "sendrecv"

request ANY sdp-header Audio-Connection-Info modify "0.0.0.0" "184.71.242.2"
request ANY sdp-header Connection-Info modify "0.0.0.0" "184.71.242.2"
response ANY sdp-header Connection-Info modify "0.0.0.0" "184.71.242.2"
response ANY sdp-header Audio-Connection-Info modify "0.0.0.0" "184.71.242.2"

If that didn't work then i think i have to wait till TAC's answer on this.

Anthony Brandelli · ‎10-19-2014

Added the four new lines to the same voice class, unfortunately this did not have an effect.

Thanks for the help, let me know if there is anything else I can provide.

Anthony

Manish Prasad · ‎10-19-2014

This certainly will be my last try :)

Can you please provide packet capture for inside and outside interface of CUBE router. I just want to check whether RTP stream has reached its interface or not.

Anthony Brandelli · ‎10-19-2014

Attached is one pcap file from both interfaces on the router.

Thanks,

Anthony

Manish Prasad · ‎10-20-2014

Here is the summary of packet capture

ITSP OK
Connection Information (c): IN IP4 204.101.100.12
Media Description, name and address (m): audio 11750 RTP/AVP 0 101
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): ptime:20

CUBE ACK
Connection Information (c): IN IP4 184.71.242.2
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 18780 RTP/AVP 0 101
Connection Information (c): IN IP4 184.71.242.2
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000

---------------------------------------------------------------------------------------
CUBE OK
Media Description, name and address (m): audio 18782 RTP/AVP 0 101
Connection Information (c): IN IP4 172.18.0.1
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): ptime:20

CUCM ACK
Connection Information (c): IN IP4 10.135.191.12
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 24608 RTP/AVP 0
Media Attribute (a): X-cisco-media:umoh
Media Attribute (a): ptime:20
Media Attribute (a): rtpmap:0 PCMU/8000

And this is actual RTP packet exchange between CUCM , CUBE and ITSP.

Src: 10.135.191.12 (10.135.191.12), Dst: 172.18.0.1 (172.18.0.1)
User Datagram Protocol, Src Port: 24608 (24608), Dst Port: 18782 (18782)

Src: 184.71.242.2 (184.71.242.2), Dst: 204.101.100.12 (204.101.100.12)
User Datagram Protocol, Src Port: 18780 (18780), Dst Port: 11750 (11750)

Every thing is matching from source to destination , i don't know what stopping the media to play at caller phone. Two possibility that i can think of right now first ANN file is corrupt (which is not possible as in other cases it playing correctly) and second there is something ahead of CUBE which is blocking this media.

When you receive an incoming call from ITSP in normal case scenario, do you hear ringback or moh ? Can you share packet capture from CUBE for a moh call ?