Solved: Re: CME SIP Trunk no ringback incoming pstn calls

chevymannie · ‎05-27-2012

I'm in the process of testing a sip trunk for a cutover and I've been having an issue with ringback. When a user dials in from the PSTN, the IP Phone rings, but no ringback is heard by the PSTN caller. If we dial outgoing to the PSTN, everything works fine.

Ayodeji Okanlawon · ‎06-04-2012

After much investigation here is what I have concluded is going on..

First of all we need to understand how ringback is played with SIP..

1. if a 180 (Ringing) has been received but there are no incoming
media packets (i.e 180 without SDP), generate local ringing.

2. If "183 Session Progress" with SDP is received it will then expect to receive the

ringback tone as RTP packets from the remote server, and generates no ringback tone.

So in your scenario, 180 ringing without SDP is sent to your provider, this implies that according to RFC 3960 your provider should generate its own local ringback. But this is not happening..

Now what I think is going on is this, your provider wants SDP in your 180 ringing so that ringback is heard from your end or they want you to send a 183 with SDP. This is also acceptable in RFC 3960.

So this is not a CCME problem like I have said before and the traces show.

You need to speak with your provider to know why they are not generating their own ringback. Or you need to find away to send 180 with SDP to your provider as this is what they can accept.. I am not sure I know how to configure CCME to send 180 with SDP or send 183 with SDP..You can ask on the forum..but I strongly suggest yo speak with your ITSP first.

Can you confirm on the outbound call if your provider sends 180 with SDP or 183 or just 180...

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View solution in original post

Ayodeji Okanlawon · ‎05-28-2012

SIP Trunks use Annunciator to play ringback. Do you have MRGL with an ANNunicator device in it. Have you also assigned the MRGL to the the SIP trunk

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paolo bevilacqua · ‎05-28-2012

Aok,

OP has CME, not CM. He can search for the "troubleshooting ringback" document.

Ayodeji Okanlawon · ‎05-28-2012

Paolo,

Thanks I totally missed that!

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chevymannie · ‎06-01-2012

No joy on that. Still not getting any ringback to the PSTN. Works fine outgoing.

Ayodeji Okanlawon · ‎06-01-2012

Can you send a

debug ccsip messages?

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chevymannie · ‎06-01-2012

Jun 1 21:14:36.590: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:601XXXXXXX@X.X.X.X:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDPX.X.X.X:5060;rport;branch=z9hG4bK-756e4951aeac6ba45e4b3d1a7e35aac8-X.X.X.X-1

Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info

Max-Forwards: 70

Call-ID: 669F14D5@X.X.X.X

From: <601XXXXXXX8>;tag=X.X.X.X+1+33cfa6+b6677e13;isup-oli=61

To: <601XXXXXXX>

CSeq: 948760796 INVITE

Expires: 180

Organization:

Supported: 100rel

Content-Length: 167

Content-Type: application/sdp

Contact: <601XXXXXXX>;isup-oli=61

P-Asserted-Identity: <601XXXXXXX>

v=0

o=- 2849730703 2849730703 IN IP4 X.X.X.X

s=-

c=IN IP4 X.X.X.X

t=0 0

m=audio 33590 RTP/AVP 2 0 101

a=rtpmap:101 telephone-event/8000

a=ptime:20

Jun 1 21:14:36.598: //21314/A038A84A97B6/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP X.X.X.X:5060;rport;branch=z9hG4bK-756e4951aeac6ba45e4b3d1a7e35aac8-X.X.X.X-1

From: <601XXXXXXX>;tag=X.X.X.X+1+33cfa6+b6677e13;isup-oli=61

To: <601XXXXXXX>

Date: Fri, 01 Jun 2012 21:14:36 GMT

Call-ID: 669F14D5@X.X.X.X

CSeq: 948760796 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Jun 1 21:14:36.606: //21314/A038A84A97B6/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP X.X.X.X:5060;rport;branch=z9hG4bK-756e4951aeac6ba45e4b3d1a7e35aac8-X.X.X.X-1

From: <601XXXXXXX>;tag=X.X.X.X+1+33cfa6+b6677e13;isup-oli=61

To: <601XXXXXXX>;tag=244DEEC0-361

Date: Fri, 01 Jun 2012 21:14:36 GMT

Call-ID: 669F14D5@X.X.X.X

CSeq: 948760796 INVITE

Require: 100rel

RSeq: 58

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

Allow-Events: telephone-event

Contact: <601XXXXXXX>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Jun 1 21:14:36.654: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

PRACK sip:

601XXXXXXX@X.X.X.X:5060:5060 SIP/2.0

Via: SIP/2.0/UDPX.X.X.X:5060;branch=z9hG4bK-bb4437ef8a5c5a08899d7bf8067b41e5-X.X.X.X-1

Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info

Max-Forwards: 70

Call-ID: 669F14D5@X.X.X.X

From: <601XXXXXXX>;tag=X.X.X.X+1+33cfa6+b6677e13;isup-oli=61

To: <601XXXXXXX>;tag=244DEEC0-361

CSeq: 948760797 PRACK

RAck: 58 948760796 INVITE

Organization:

Supported: 100rel

Content-Length: 0

Jun 1 21:14:36.654: //21314/A038A84A97B6/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK-bb4437ef8a5c5a08899d7bf8067b41e5-208.149.73.5-1

From: <601XXXXXXX>;tag=X.X.X.X+1+33cfa6+b6677e13;isup-oli=61

To: <601XXXXXXX>;tag=244DEEC0-361

Date: Fri, 01 Jun 2012 21:14:36 GMT

Call-ID: 669F14D5@X.X.X.X

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 948760797 PRACK

Content-Length: 0

Ayodeji Okanlawon · ‎06-01-2012

Hi,

I have looked at the logs and I cant see anything abnormal...But to look further can you send

debug ccsip all

debug voip ccapi inout..

This will let us see if you are sending alert and ringback to the PSTN

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chevymannie · ‎06-01-2012

Here you go. Hope I got all the output.

Ayodeji Okanlawon · ‎06-01-2012

Hi,

I have looked at your the ccapi log and here we find what we are looking for..

++++The outbound leg:Call id 21440+++ sends an alert to the gateway via CCAPI

un 1 21:43:19.721: //21440/A2CC97DA9881/CCAPI/cc_api_call_proceeding:
   Interface=0x400A9694, Progress Indication=NULL(0)
Jun 1 21:43:19.725: //21440/A2CC97DA9881/CCAPI/cc_api_call_alert:
   Interface=0x400A9694, Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Jun 1 21:43:19.725: //21440/A2CC97DA9881/CCAPI/cc_api_call_alert:
   Call Entry(Retry Count=0, Responsed=TRUE)

++++CCAPI then sends the alert to CUCM on the inbound leg callid:21439+++++

Jun 1 21:43:19.725: //21439/A2CC97DA9881/CCAPI/ccCallAlert:
Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Jun 1 21:43:19.725: //21439/A2CC97DA9881/CCAPI/ccCallAlert:
Call Entry(Responsed=TRUE, Alert Sent=TRUE)

So we can see that the ringback is been sent to CUCM by the far end.

Can you send a copu of your config with your dial-peers section. Do you have any service configured on the dial-peer?

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Ayodeji Okanlawon · ‎06-01-2012

Chev,

I think you may have sent the wrong debug details for the voip ccapi inout. Because your sip debugs look different in terms of the calling number and called number...Can you please confirm that this is the right debug..

your sip debug shows that you are sending ringing and 200 ok at the same time..hence why the PSTN caller does not hear any ringing..

Jun 1 21:46:00.581: //21456/02CBC92298AA/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 180 Ringing

Jun 1 21:46:00.581: //21456/02CBC92298AA/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK Jun 1 21:46:00.581: //21456/02CBC92298AA/SIP/Msg/ccsipDisplayMsg:

Please send a sh run of your router ...

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chevymannie · ‎06-03-2012

I noticed that from the debugs after I looked at another post. Do you know of a way to delay both being sent at the same time.

Here's the show run.

chevymannie · ‎06-03-2012

I just disabled early media offer under sip-ua mode. It delays the 200 ok after the 180 trying for a few miliseconds, but still no ringback.

Ayodeji Okanlawon · ‎06-04-2012

can you send a debug voip ccapi inout only.

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chevymannie · ‎06-04-2012

Here it is. Hope you can find something, I'm stumped.