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Replies

CUBE Rings first then gives busy tone

sarwarm123
Level 1
Level 1

We have recently configured SIP trunk in our organization. However we noticed if some PSTN number is out of order or busy it rings first then give you busy tone. I think it is a SIP generated ring tone. Can somebody help me to resolve this issue

Call Flow

IP-Phone--->CUCM--->SIP--->CUBE---->SIP---->ITSP

2 Accepted Solutions

Accepted Solutions

1. You are not gettting early media on 180..This wont help you here...You are getting 183 with SDP..totally different

2. Same answer as above since you are getting 183 with SDP not 180 with SDP

3...Look at your 183 you are getting fom ITSP..

Received:

SIP/2.0 183 Session Progress

Require: 100rel

Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEA10B3

RSeq: 1

To: <02476304888>;tag=3575177498-69142

From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C

Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98

CSeq: 101 INVITE

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

Contact: <02476304888>

Content-Type: application/sdp

Accept: application/sdp

Content-Length: 209

v=0

o=MSX10 3588387992081225328 1 IN IP4 10.128.0.1

s=sip call

c=IN IP4 10.128.0.2

t=0 0

m=audio 34706 RTP/AVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

497611: Apr 17 08:51:38.076: //9913/021838000002/SIP/Msg/ccsipDisplayMsg:

4..Yes, this is how CUCM plays ringback locally..If it receives 183 without SDP it will tell the phone to play ringback..If it receives 183 with SDP even if there is no media in the SDP then it will still cut through and try to listen to what the provider is sending......If your provider doesnt want to send any media in 183 then they should send 183 without SDP..Although it is perfectly normal as speficied in the RFC that they can send 183 with SDP without sending any media but CUCM cant tell if there is media or not

5.  If ITSP sends 183 with SDP, ITSP has the responsibility to play the ringback or any audio, for the CUBE or CUCM, there is only one thing to do which is to cut through media,  by theory, the device along the way should not change it from 183 with SDP to 180 which will change the behavior and might cause further issue.

If you really want to change the behavior, you can use Lua script in CUCM version later than 8.5(1)  to change the 183 message to 180, which might force the CUCM to play the local ringback. The feature in CUCM called SIP transparency and normalization.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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View solution in original post

If you have changed your SIP trunk to h323 gateway in CUCM, then you need to change the dial-peer incoming from cucm and outgoing to cucm to h323 dial-peer..From the logs you are still using sip protocol on the dial-peer...

You also need to bear in mind that your CUBE is sending EO to cucm..with h323 gateway you will need to enable inboud and outbound fast start on the gateway...For this you need an MTP device...

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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View solution in original post

41 Replies 41

daniel.bloom
Level 1
Level 1

Can you past the output of "debug ccsip messages". This should tell us which device is signalling the ringtone.

Sent from Cisco Technical Support iPad App

Please see ccsip messages debug

Received:
INVITE sip:902476304888@10.116.143.151:5060 SIP/2.0
Via: SIP/2.0/TCP 10.102.243.13:5060;branch=z9hG4bKcb5c55637e8d4
From: "Muhammad Sarwar" <44043>;tag=4196987~867df168-c5c9-4b90-8b3a-75c7400ce77b-45118968
To: <902476304888>
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: 2183800-16e16299-84f54-df3660a@10.102.243.13
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <10.102.243.13:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0035141632-0000065536-0000140420-0234055178
Session-Expires:  1800
P-Asserted-Identity: "Muhammad Sarwar" <44043>
Remote-Party-ID: "Muhammad Sarwar" <44043>;party=calling;screen=yes;privacy=off
Contact: <44043>
Max-Forwards: 70
Content-Length: 0


497603: Apr 17 08:51:37.964: //9912/021838000002/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:02476304888@10.102.243.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.116.143.151:5060;branch=z9hG4bK3EE9122
Remote-Party-ID: "Muhammad Sarwar" <44043>;party=calling;screen=yes;privacy=off
From: "Muhammad Sarwar" <44043>;tag=46277C84-6DC
To: <02476304888>
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: D979A596-A67211E2-A3C6B6AF-278B3D2@10.116.143.151
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0035141632-0000065536-0000140420-0234055178
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1366188697
Contact: <44043>
Expires: 180
Allow-Events: telephone-e
CDC-CUBE#vent
Max-Forwards: 69
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 252

v=0
o=CiscoSystemsSIP-GW-UserAgent 907 5295 IN IP4 10.116.143.151
s=SIP Call
c=IN IP4 10.116.143.151
t=0 0
m=audio 32706 RTP/AVP 8 101
c=IN IP4 10.116.143.151
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

497604: Apr 17 08:51:37.964: //9911/021838000002/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.102.243.13:5060;branch=z9hG4bKcb5c55637e8d4
From: "Muhammad Sarwar" <44043>;tag=4196987~867df168-c5c9-4b90-8b3a-75c7400ce77b-45118968
To: <902476304888>
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: 2183800-16e16299-84f54-df3660a@10.102.243.13
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Content-Length: 0


497605: Apr 17 08:51:37.968: //9912/021838000002/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.116.143.151:5060;branch=z9hG4bK3EE9122
From: "Muhammad Sarwar" <44043>;tag=46277C84-6DC
To: <02476304888>
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: D979A596-A67211E2-A3C6B6AF-278B3D2@10.116.143.151
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0


497606: Apr 17 08:51:37.972: //9912/021838000002/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.116.143.151:5060;branch=z9hG4bK3EE9122
From: "Muhammad Sarwar" <44043>;tag=46277C84-6DC
To: <02476304888>;tag=4196988~867df168-c5c9-4b90-8b3a-75c7400ce77b-45118969
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: D979A596-A67211E2-A3C6B6AF-278B3D2@10.116.143.151
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=1
Content-Length: 0


497607: Apr 17 08:51:37.976: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:02476304888@10.102.243.13:5060 SIP/2.0
Via: SIP/2.0/UDP 10.116.143.151:5060;branch=z9hG4bK3EE9122
From: "Muhammad Sarwar" <44043>;tag=46277C84-6DC
To: <02476304888>;tag=4196988~867df168-c5c9-4b90-8b3a-75c7400ce77b-45118969
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: D979A596-A67211E2-A3C6B6AF-278B3D2@10.116.143.151
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


497608: Apr 17 08:51:37.976: //9913/021838000002/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:02476304888@10.128.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEA10B3
Remote-Party-ID: "Muhammad Sarwar" <44043>;party=calling;screen=no;privacy=full
From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C
To: <02476304888>
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0035141632-0000065536-0000140420-0234055178
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1366188697
Contact: <44043>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 243

v=0
o=CiscoSystemsSIP-GW-UserAgent 933 7445 IN IP4 10.60.34.98
s=SIP Call
c=IN IP4 10.60.34.98
t=0 0
m=audio 32708 RTP/AVP 8 101
c=IN IP4 10.60.34.98
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

497609: Apr 17 08:51:37.984: //9913/021838000002/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEA10B3
From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C
To: <02476304888>
Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98
CSeq: 101 INVITE
Timestamp: 1366188697
Content-Length: 0


497610:
CDC-CUBE# Apr 17 08:51:38.076: //9913/021838000002/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Require: 100rel
Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEA10B3
RSeq: 1
To: <02476304888>;tag=3575177498-69142
From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C
Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98
CSeq: 101 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: <02476304888>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 209

v=0
o=MSX10 3588387992081225328 1 IN IP4 10.128.0.1
s=sip call
c=IN IP4 10.128.0.2
t=0 0
m=audio 34706 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

497611: Apr 17 08:51:38.076: //9913/021838000002/SIP/Msg/ccsipDisplayMsg:
Sent:
PRACK sip:02476304888@10.128.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEB2590
From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C
To: <02476304888>;tag=3575177498-69142
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98
CSeq: 102 PRACK
RAck: 1 101 INVITE
Allow-Events: telephone-event
Max-Forwards: 70
Content-Length: 0


497612: Apr 17 08:51:38.080: //9911/021838000002/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.102.243.13:5060;branch=z9hG4bKcb5c55637e8d4
From: "Muhammad Sarwar" <44043>;tag=4196987~867df168-c5c9-4b90-8b3a-75c7400ce77b-45118968
To: <902476304888>;tag=46277CF4-1919
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: 2183800-16e16299-84f54-df3660a@10.102.243.13
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <02476304888>;party=called;screen=yes;privacy=off
Contact: <902476304888>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 277

v=0
o=CiscoSystemsSIP-GW-UserAgent 7861 6742 IN IP4 10.116.143.151
s=SIP Call
c=IN IP4 10.116.143.151
t=0 0
m=audio 32704 RTP/AVP 8 101 19
c=IN IP4 10.116.143.151
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20

497613: Apr 17 08:51:38.088: //9913/021838000002/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEB2590
To: <02476304888>;tag=3575177498-69142
From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C
Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98
CSeq: 102 PRACK
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: <02476304888>
Content-Length: 0


497614: Apr 17 08:51:40.220: //9913/021838000002/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEA10B3
To: <02476304888>;tag=3575177498-69142
From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C
Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98
CSeq: 101 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: <02476304888>
Reason: Q.850;cause=17
Content-Length: 0


497615: Apr 17 08:51:40.220: //9913/021838000002/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:02476304888@10.128.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEA10B3
From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C
To: <02476304888>;tag=3575177498-69142
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


497616: Apr 17 08:51:40.220: //9911/021838000002/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 486 Busy here
Via: SIP/2.0/TCP 10.102.243.13:5060;branch=z9hG4bKcb5c55637e8d4
From: "Muhammad Sarwar" <44043>;tag=4196987~867df168-c5c9-4b90-8b3a-75c7400ce77b-45118968
To: <902476304888>;tag=46277CF4-1919
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: 2183800-16e16299-84f54-df3660a@10.102.243.13
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Reason: Q.850;cause=17
Content-Length: 0


497617: Apr 17 08:51:40.224: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:

CDC-CUBE#ACK sip:902476304888@10.116.143.151:5060 SIP/2.0
Via: SIP/2.0/TCP 10.102.243.13:5060;branch=z9hG4bKcb5c55637e8d4
From: "Muhammad Sarwar" <44043>;tag=4196987~867df168-c5c9-4b90-8b3a-75c7400ce77b-45118968
To: <902476304888>;tag=46277CF4-1919
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: 2183800-16e16299-84f54-df3660a@10.102.243.13
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

You are receiving session progress from the host 10.128.0.1 as per the below section of the debug. This is when the ring back tone would be heard.

Received:
SIP/2.0 183 Session Progress
Require: 100rel
Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEA10B3
RSeq: 1
To: <02476304888>;tag=3575177498-69142
From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C
Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98
CSeq: 101 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: <02476304888>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 209
v=0
o=MSX10 3588387992081225328 1 IN IP4 10.128.0.1
s=sip call
c=IN IP4 10.128.0.2
t=0 0
m=audio 34706 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
497611: Apr 17 08:51:38.076: //9913/021838000002/SIP/Msg/ccsipDisplayMsg:

Sent:
PRACK sip:02476304888@10.128.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEB2590
From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C
To: <02476304888>;tag=3575177498-69142
Date: Wed, 17 Apr 2013 08:51:37 GMT
Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98
CSeq: 102 PRACK
RAck: 1 101 INVITE
Allow-Events: telephone-event


Shortly after that you get a busy signal from the same host 10.128.0.1

Received:
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEA10B3
To: <02476304888>;tag=3575177498-69142
From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C
Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98
CSeq: 101 INVITE
Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: <02476304888>
Reason: Q.850;cause=17
Content-Length: 0

Sent from Cisco Technical Support iPad App

Hi Daniel.

Thanks for your reply. 10.128.0.1 is a service provider. How could we resolve this problem this, is that service provider needs to do something or we need to do something?

This is generated on the service provider side so nothing you can do.

Sent from Cisco Technical Support iPhone App

Daniel

Thanks mate, I am going to contact Service Provider, see what they reply back

Muhammad,

First of all (+5) to Daniel...Great Job buddy..

This is a common problem with most sip providers even here in the UK..Another scenario where we see this is when you make a call and the other party rejects the call, your phone rings once or twice before you get a busy tone.

You can do this test also and confirm this behaviour...

SOme people say its "fake ring" Looks like this providers wants to give some form of session progress and the only to do it is to provide a fake ring...

Lets know what you find

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts

Hi Aok,

I got following reply from the service provider, is that 180 ringing casuing the issue?

“I’ve attached the trace of those calls from our side. Gamma doesn’t send any ringing to site within the media stream, or signal 180 Ringing in the SIP, so the customers kit must be playing its own local ringing. There may be a config option to prevent local ringing in this scenario.”

Muhammad,

Looking at the trace again, I can see this..

The call was first of all sent to 10.102.243.13. This device didnt send any ringing but a 404 not found after trying..

Sent:

INVITE sip:02476304888@10.102.243.13:5060 SIP/2.0

Via: SIP/2.0/UDP 10.116.143.151:5060;branch=z9hG4bK3EE9122

Remote-Party-ID: "Muhammad Sarwar"

Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.116.143.151:5060;branch=z9hG4bK3EE9122

The gateway then sent the call to a second device..10.128.0.1 and it is this device that is sending 183 with SDP

Sent:

INVITE sip:02476304888@10.128.0.1:5060 SIP/2.0

Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEA10B3

Received:

SIP/2.0 183 Session Progress

Require: 100rel

Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK3EEA10B3

RSeq: 1

To: <02476304888>;tag=3575177498-69142

From: "Muhammad Sarwar" <44043>;tag=46277C8C-1B9C

Call-ID: D97ADDE6-A67211E2-A3C8B6AF-278B3D2@10.60.34.98

The question is which trace is your ITSP guy looking at? This is definitely from his device..You can send this log to him. The device do not behave in the same way, once doesnt send anything, the other is sending session progress. Logs do  not lie..send them to him and let him earn his wages

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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Hi Aok,

He looked at wireshark traces, I have attached the traces for your consideration. When I looked the traces I can also see from the source (ITSP) 183 session progress is generated. I don't know why he didn't notice that.

I am going to send Cisco traces to him as well, so he can earn his wages

I have one question

What is difference between Signaling 180 and 183, if you see his reply he was emphasizing on the signaling 180 not the 183. Does 183 and 180 same thing?

Hi Aok,

I just collected the another debug logs, but dont see 404 not found However I can see SIP/2.0 486 Busy here Please see attached debug log for a moment

The reason you don't see the "404 not found" message in the latest logs is because your CUBE router is not sending the first INVITE to 10.102.243.13 like it does in the first set of debugs.

Sent:

INVITE sip:02476304888@10.102.243.13:5060 SIP/2.0

Via: SIP/2.0/UDP 10.116.143.151:5060;branch=z9hG4bK3EE9122

Remote-Party-ID: "Muhammad Sarwar" <44043>;party=calling;screen=yes;privacy=off

From: "Muhammad Sarwar" <44043>;tag=46277C84-6DC

To: <02476304888>

Date: Wed, 17 Apr 2013 08:51:37 GMT

Call-ID:

D979A596-A67211E2-A3C6B6AF-278B3D2@10.116.143.151

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

Min-SE:  1800

Cisco-Guid: 0035141632-0000065536-0000140420-0234055178

User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2

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

CSeq: 101 INVITE

Timestamp: 1366188697

Contact: <44043>

Expires: 180

Allow-Events: telephone-e

CDC-CUBE#vent

Max-Forwards: 69

Session-Expires:  1800

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 252

v=0
o=CiscoSystemsSIP-GW-UserAgent 907 5295 IN IP4 10.116.143.151
s=SIP Call
c=IN IP4 10.116.143.151
t=0 0
m=audio 32706 RTP/AVP 8 101
c=IN IP4 10.116.143.151
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

497604: Apr 17 08:51:37.964: //9911/021838000002/SIP/Msg/ccsipDisplayMsg:


This looks like it might have been an error in your configuration as the host 10.102.243.13 appears to be your CUCM 8.6 server. The latest set of debugs does not try and route the call to this server however sends it directly to the ITSP.

Sent:

INVITE sip:07776137568@10.128.0.1:5060 SIP/2.0

Via: SIP/2.0/UDP 10.60.34.98:5060;branch=z9hG4bK5CA81E2F

From: <44043>;tag=52B794B0-AB7

To: <07776137568>

Date: Fri, 19 Apr 2013 19:24:47 GMT

Call-ID: A196A647-A85D11E2-8900B6AF-278B3D2@10.60.34.98

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

Min-SE:  1800

Cisco-Guid: 3401166592-0000065536-0000148451-0234055178

User-Agent: Cisco-SIPGateway/IOS-15.2.4.M2

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

CSeq: 101 INVITE

Timestamp: 1366399487

Contact: <44043>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 68

P-Asserted-Identity: <44043>

Session-Expires:  1800

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 244

Privacy: id

v=0

o=CiscoSystemsSIP-GW-UserAgent 1564 9925 IN IP4 10.60.34.98

s=SIP Call

c=IN IP4 10.60.34.98

t=0 0

m=audio 22714 RTP/AVP 8 101

c=IN IP4 10.60.34.98

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

You are still hearing a ring back tone because the ITSP is providing session progress.

The 183 message is used by the called gateway to provide media to the calling gateway prior to the call being completed. This media can be used to play anouncements, ringback or music to the calling gateway prior to the call actually being completed. An example of this is when you call a mobile phone user and hear a message similar to "this number is not available at this time" from the mobile phone carrier. You need to be able to hear this message prior to the call completing so that the call is not billed. Some ITSPs also use the 183 session progress to generate ringback tone at the called gateway rather than letting the calling gateway generate the ringback tone (like what we see with your debugs). This is where the 180 Ringing message is used. When the called gateway wants the calling gateway to generate the ringback tone it will send a 180 Ringing message.

Daniel, (+5) excellent explanation..

Just add a litte note to it..When CUBE receives 183 with SDP, it cuts through to listen  whatever media the far end is sending (this is what is called early media). When CUBE gets 180 with ringing (without SDP) CUCM tells the ip phone to generate ringback locally.

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

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What a great explanation by both of you Aok and Daniel and thanks to clear my concept about messages 180 and 183. Its time to go back to ITSP