CME no ringback tone for outgoing calls over SIP Trunk

[Hassan Ali Hassan Ali Gazal]

I have a CME Router 2921 connected to SIP Router "Peer to Peer", which is giving me VOIP Service. i can make inbound calls normal and i can hear ringback tone when tested incoming calls from my Mobile or any landline Phone.

Issue is when i try to make outbound call to my mobile or any other phone i can see the phone ringing but i can't hear any ringback tone.

below in my configuration and Debug Cssip message for outgoing calls.

!
!
!
voice service voip
ip address trusted list
ipv4 10.200.200.50
ipv4 10.200.200.51
ipv4 10.200.200.243
ipv4 10.200.200.253
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711ulaw
h323
sip
bind control source-interface GigabitEthernet0/0.20
bind media source-interface GigabitEthernet0/0.20
registrar server expires max 600 min 60
no silent-discard untrusted
g729 annexb-all
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
!
!
voice class custom-cptone KW
dualtone disconnect
frequency 425
cadence 320 320
!
!
!
!
voice translation-rule 1
rule 1 /^8/ //
!
voice translation-rule 2
rule 1 /10./ /22066618/
rule 2 /11./ /22066619/
rule 3 /12./ /22066620/
rule 4 /13./ /22066621/
rule 5 /14./ /22066619/
rule 6 /19./ /22066619/
!
voice translation-rule 3
rule 1 /^9/ //
!
voice translation-rule 4
rule 1 /22066618/ /104/
rule 2 /22066619/ /104/
rule 3 /22066620/ /104/
rule 4 /22066621/ /104/
!
!
voice translation-profile INCOMING_CALLS
translate called 4
!
voice translation-profile KSA_OUTGOING
translate called 1
!
voice translation-profile OUTGOING_CALLS
translate calling 2
translate called 3
!
!
!

!
interface GigabitEthernet0/0.10
encapsulation dot1Q 10
ip address 192.168.10.252 255.255.255.0
!
interface GigabitEthernet0/0.20
encapsulation dot1Q 20
ip address 192.168.20.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
!
interface GigabitEthernet0/1
description PBX-LAN-IP
ip address 10.108.42.2 255.255.255.0
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
!

!
!
voice-port 0/3/0
no supervisory disconnect lcfo
no battery-reversal
connection plar 22066617
description FAX-MACHINE
caller-id enable
!
voice-port 0/3/1
!
voice-port 0/3/2
!
voice-port 0/3/3
!
!
!
!
!
dial-peer voice 2 voip
description Mobile
translation-profile outgoing OUTGOING_CALLS
destination-pattern 9[569].......$
session protocol sipv2
session target ipv4:10.200.200.50
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 3 voip
description HotLines
translation-profile outgoing OUTGOING_CALLS
destination-pattern 9[1]......$
session protocol sipv2
session target ipv4:10.200.200.50
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 4 voip
description INCOMING-CALLS
translation-profile incoming INCOMING_CALLS
destination-pattern 104
session protocol sipv2
session target ipv4:10.200.200.50
incoming called-number ^220666..$
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 10 pots
destination-pattern 22066617$
port 0/3/0
forward-digits 0
!
dial-peer voice 5 voip
description INCOMING-FAX
destination-pattern ^22066617$
session protocol sipv2
session target ipv4:10.200.200.50
incoming called-number ^22066617$
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no vad
!
dial-peer voice 6 voip
description Landline
translation-profile outgoing OUTGOING_CALLS
destination-pattern 9[2].......$
session protocol sipv2
session target ipv4:10.200.200.50
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
!
!
!
gatekeeper
shutdown
!
!
telephony-service
max-ephones 100
max-dn 100
ip source-address 192.168.20.1 port 2000
system message KPC CO. W.L.L
load 7911 SCCP11.9-2-1S
load 7912 CP7912080004SCCP080108A
load 7941 SCCP41.9-2-1S
load 7961 SCCP41.9-2-1S
time-zone 31
date-format dd-mm-yy
max-conferences 8 gain -6
moh enable-g711 "flash:en_bacd_music_on_hold.au"
web admin system name admin password CISCO@123
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Apr 12 2018 18:28:43
!
!

[Jaca]

Try adding to the matched VoIP dial peer:

progress_ind setup enable 3

and

progress_ind alert enable 8

to the POTS dial peer.

This may not solve your issue, but if it does or you do end up finding a solution test call transfers to make sure you can hear ring back on a transfer.

[Ayodeji Okanlawon · Accepted solution]

(+5) Nipun,

This is not a CME issue. By default CME will generate ring back locally if SDP is not received in either 180 or 183 response. If SDP is included in the 180 or 183 response, instead of playing ring back locally, CME will connect media and play whatever the called device is sending such as ring back, busy or announcement. In this this looks like the device that you have called is including SDP in its 180/183 but is not sending any media before the 200 OK ( that media should ideally be your ring back unless they want to play something else to you)

So go back to them and let them sort this out

[R0g22]

This is what I see in the logs -

1. Your ITSP sends a 183/w SDP with a 100 rel -

Apr 22 16:17:47.075: //70/862B783F8077/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 10.108.42.2:1036;branch=z9hG4bK151D9D

From: "199" <sip:22066619@10.108.42.2:1036>;tag=140E30-2137

To: <sip:99109912@10.200.200.50>;tag=6FA1270-22F7

Date: Sun, 22 Apr 2018 16:19:05 GMT

Call-ID: 873FF656-457F11E8-807C9919-BDA63A5E@192.168.20.1

Timestamp: 1524413866

CSeq: 101 INVITE

Require: 100rel

RSeq: 5353

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

Allow-Events: kpml, telephone-event

Contact: <sip:99109912@10.200.200.243:5060>

Record-Route: <sip:10.200.200.50;lr;did=15f.72253dc5>

Call-Info: <sip:10.200.200.243:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

Server: Cisco-SIPGateway/IOS-12.x

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 246

v=0

o=CiscoSystemsSIP-GW-UserAgent 2828 2838 IN IP4 10.200.200.243

s=SIP Call

c=IN IP4 10.200.200.243

t=0 0

m=audio 18770 RTP/AVP 0 100

c=IN IP4 10.200.200.243

a=rtpmap:0 PCMU/8000

a=rtpmap:100 X-NSE/8000

a=fmtp:100 192-194

a=ptime:20

2. CME does a PRACK correctly -

Apr 22 16:17:47.083: //70/862B783F8077/SIP/Msg/ccsipDisplayMsg:
Sent:
PRACK sip:99109912@10.200.200.243:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK1610A8

From: "199" <sip:22066619@192.168.20.1>;tag=140E30-2137

To: <sip:99109912@10.200.200.50>;tag=6FA1270-22F7

Date: Sun, 22 Apr 2018 16:17:46 GMT

Call-ID: 873FF656-457F11E8-807C9919-BDA63A5E@192.168.20.1

CSeq: 102 PRACK

RAck: 5353 101 INVITE

Route: <sip:10.200.200.50;lr;did=15f.72253dc5>

Allow-Events: telephone-event

Max-Forwards: 70

Content-Length: 0

3. ITSP sends a 200 OK for PRACK -

Apr 22 16:17:47.115: //70/862B783F8077/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.108.42.2:1037;branch=z9hG4bK1610A8

From: "199" <sip:22066619@10.108.42.2:1036>;tag=140E30-2137

To: <sip:99109912@10.200.200.50>;tag=6FA1270-22F7

Date: Sun, 22 Apr 2018 16:19:06 GMT

Call-ID: 873FF656-457F11E8-807C9919-BDA63A5E@192.168.20.1

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 102 PRACK

Content-Length: 0

At this point there should be ring back in the form of early media but your ITSP keeps sending 183 again. IOS does a PRACK etc. This happens 3 times in total and then there is a 200 OK for the initial INVITE when the call actually answers. I am not sure why your ITSP keeps sending 183, they do get a PRACK because they send a 200 OK for it so that part should be good.

I am assuming the following is your external facing interface and a firewall at the edge is taking care of NAT for you -

interface GigabitEthernet0/0.20
encapsulation dot1Q 20
ip address 192.168.20.1 255.255.255.0
ip nat inside
ip virtual-reassembly in

If that is correct, contact to your ITSP with the logs. They need to look into it.

[Hassan Ali Hassan Ali Gazal]

Thanks all, issue has been resolved. the issue which am facing now is i have FAX machine connected to FXS Card, when i dial the FAX number from my mobile its ringing and then deeply silent.

[Ayodeji Okanlawon · Accepted solution]

And what was the resolution? You didn't provide feed back, neither did you rate posts and then you come to ask for more help...

[Hassan Ali Hassan Ali Gazal]

issue was from ISP Side, he sent the ringback tone from his side.