CUBE Rings first then gives busy tone

[sarwarm123]

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

[sarwarm123]

Hi Aok,

I had detailed conversation with ITSP but they are still insisting that they are not sending audio in response 183. Before we go to Cisco TAC I wanted to make sure settings at our end are ok. I have few questions in my mind will you please clarify that.

1) Do I need to check Disable Early Media on 180 on SIP profile (Currently it is uncheck)

2) Do we need to apply command in CUBE Disable Early Media 180 (currently this command has not applied)

3) How could I check service provider sending the response 183 with SDP or without SDP

4) I read Cisco doc if we receive response 183 without SDP CUCM will generate local ringback. However if it comes with SDP but no media then we don't hear ringback so to resolve this issue check Disable Early Media on 180

5) Is any thing we can do to disable or modify resoponse 183 to disable local ringback if we are getting 183 without SDP from service provider

According to ITSP we are getting response 183 (not sure its sdp or without sdp) with no media. So what do you think where we stand?

[Ayodeji Okanlawon · Accepted solution]

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.

Please rate all useful posts

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

[sarwarm123]

Hi Guys,

I had a chat with ITSP and they confirmed the are sending 183 with SDP but without media except few RTP packets. They are not willing to do 183 without SDP or add media.

Please see below their response in writing

"Just to confirm what we discussed on our call. Gamma will always send a 183 with SDP when the call leaves our PSTN gateways. This is to ensure the calling device has media parameters to listen for early media should the far-end send it."

Can we now try to change the behavior by using Lua script, can you please help in that please

[daniel.bloom]

There is one other option for checking which device is generating the ringback tone.

1. On the phone you are using for testing span the traffic to the computer port.
2. Attach a computer with a packet sniffer (wireshark)
3. Make a call and post the wireshark capture.

We should be able to decode the RTP stream (if it is using G.711ulaw or G.711alaw) and find which IP address is sending the ringback tone.

[sarwarm123]

Hi

I have attached wireshark file, I can see CUCM generating 180 ringing

Jabber--->CUCM----->SIP------>CUBE------->ITSP------>PSTN

[Ayodeji Okanlawon]

This packet capture you have sent is for a normal call...I am not sure what you want to get out of this..Yes CUCM is sending 180 ringing but thats totally normal..This is different from when cucm receives 183 with SDP.. I can hear audion on this call in one direction from 10.209.208 to 10.116....I dont hear anything in the other direction other than some mumbled sound...

You should do a test with the scenario we are investigating..not a normal call

Please rate all useful posts

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

[sarwarm123]

Hi Aok,

Yeah you right capture was for the normal call, sorry for that. I have attached wireshark capture of the scenario we are investigating.

As you know service provider not willing to send 183 without SDP or 183 with SDP and media. Can we not try the solution you suggested above, I mean SIP normalization script inc CUCM

[sarwarm123]

Hi Aok,

Did u get any chance to look at sip normalization script solution?

[Ayodeji Okanlawon]

You need to look at the bigger picture..This can create further problem down the line..With this solution, When your callers call destination where media is played eg announcement before 200 OK is recieved they wont be able to hear anything..Then users will start complaining again...So you need to wiegh things up, is this that important for you to break up other things....If its then I can work on something...Lua scripting isnt that simple either but I can work on it

Please rate all useful posts

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

[sarwarm123]

Hi Aok,

If there is no other solution to resolve this issue then we can take risk. It may not break other things finger cross. I would be really grateful if you can work on Lua scripting 

[Ayodeji Okanlawon]

Muhammed,

Thiking deeper about this, the lua script wont work...because when CUCM receives the 183 with SDP, we will convert it to 180 without SDP, hence cucm will tell the phome to generate ringback locally..But thats not what you want...When you call a false destination, you shouldnt get any session progress....Your provider should just send busy...Thats it

Please rate all useful posts

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

[sarwarm123]

Hi Aok,

Just wanted to inform you issue is resolved by changing it SIP trunk to H323 gateway. Outbound calls are working ok but incoming calls not giving playing ringback.

I am getting signal 180 from the service provider when we calling inbound number. Any idea why

voice service voip

ip address trusted list

  ipv4 10.X.X.X.X 255.255.255.128

   address-hiding

mode border-element

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

  header-passing

  error-passthru

  asserted-id pai

  early-offer forced

  midcall-signaling passthru

dial-peer voice 101 voip

description *** Inbound Calls from ITSP ***

translation-profile incoming STD

session protocol sipv2

session target sip-server

incoming called-number .

voice-class codec 1

dtmf-relay rtp-nte

ip qos dscp cs4 media

ip qos dscp cs3 signaling

!

dial-peer voice 2 voip

description *** Outbound LANDLINE calls to ITSP ***

destination-pattern 0[1-37]........

session protocol sipv2

session target sip-server

voice-class codec 1

voice-class sip asserted-id pai

voice-class sip profiles 4

dtmf-relay rtp-nte

ip qos dscp cs4 media

ip qos dscp cs3 signaling

no vad

!

dial-peer voice 3 voip

description *** Outbound 0845-0870 calls to ITSP ***

destination-pattern 0[8][047][0458]......

session protocol sipv2

session target sip-server

voice-class codec 1

voice-class sip asserted-id pai

voice-class sip profiles 4

dtmf-relay rtp-nte

ip qos dscp cs4 media

ip qos dscp cs3 signaling

no vad

sip-ua

no remote-party-id

disable-early-media 180

retry invite 2

retry bye 2

sip-server ipv4:10.128.0.17

[sarwarm123]

Hi Aok,

As I said above changing from SIP trunk to H323 fixed the busy tone issue but have raised another issue for the incoming call. I am getting following error. Full log has been attached, will you give your expert adivce where the problem is

SIP/2.0 503 Service Unavailable

Warning: 399 CCMSub "Unable to find a device handler for the request received on port 59733 from 10.116.153.151"

SIP/2.0 503 Service Unavailable

Warning: 399 CCMSub "Unable to find a device handler for the request received on port 59733 from 10.116.153.151"

[Ayodeji Okanlawon · Accepted solution]

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...

Please rate all useful posts

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

[sarwarm123]

You really need to look at your config..

This call is matching dial-peer 20..So you dial from cucm and send the call back to CUCM..Thats why you are not seeing any sip debugs...

Called Number=02476016423(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,

   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=20, Call Count On=FALSE,

The config on dial-peer 20 matches the dialled number

dial-peer voice 20 voip

description **** Outbound calls to Primary CUCM ***

translation-profile outgoing CLI

destination-pattern 0[1-3]........

  session target ipv4:10.102.243.13

voice-class codec 1

dtmf-relay rtp-nte

ip qos dscp cs4 media

ip qos dscp cs3 signaling