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No secondary or ring back tone on dailing any number

stanzin
Level 1
Level 1

‎08-09-2018 11:18 AM - edited ‎03-17-2019 01:19 PM

We have a sip trunk as the gateway

I am able to dial any number but especially for international calls it will take almost 15-18 sec to connect. I don't get a sec or a ring back tone.

Checked the Route plan report and all RP & TP starting with 9 has "Provide outside Dial Tone" enabled.

 

Here's a copy of the sip messages

 

Aug  9 17:19:28.458: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:90019088873090@10.74.66.5:5060 SIP/2.0
Via: SIP/2.0/TCP 10.167.131.33:5060;branch=z9hG4bK6d35f92b76b51d
From: <sip:20382@10.167.131.33>;tag=57197010~a6b8e9be-4836-49e6-b65d-1c212bf73946-117107293
To: <sip:90019088873090@10.74.66.5>
Date: Thu, 09 Aug 2018 17:19:28 GMT
Call-ID: 5d9cd580-b6c177a0-479e85-2183a70a@10.167.131.33
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM10.5
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: <sip:10.167.131.33:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED
Cisco-Guid: 1570559360-0000065536-0000180993-0562276106
Session-Expires:  1800
P-Asserted-Identity: <sip:20382@10.167.131.33>
Remote-Party-ID: <sip:20382@10.167.131.33>;party=calling;screen=yes;privacy=off
Contact: <sip:20382@10.167.131.33:5060;transport=tcp>
Max-Forwards: 70
Content-Length: 0


Aug  9 17:19:28.462: //25510/5D9CD5800002/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.167.131.33:5060;branch=z9hG4bK6d35f92b76b51d
From: <sip:20382@10.167.131.33>;tag=57197010~a6b8e9be-4836-49e6-b65d-1c212bf73946-117107293
To: <sip:90019088873090@10.74.66.5>
Date: Thu, 09 Aug 2018 18:19:28 CST
Call-ID: 5d9cd580-b6c177a0-479e85-2183a70a@10.167.131.33
CSeq: 101 INVITE
Allow-Events: kpml, telephone-event
Server: Cisco-SIPGateway/IOS-15.5.3.S6
Content-Length: 0


Aug  9 17:19:32.853: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.167.131.33:5060;branch=z9hG4bK6d35f92b76b51d
From: <sip:20382@10.167.131.33>;tag=57197010~a6b8e9be-4836-49e6-b65d-1c212bf73946-117107293
To: <sip:90019088873090@10.74.66.5>;tag=9B3BBDC-1E57
Date: Thu, 09 Aug 2018 18:19:28 CST
Call-ID: 5d9cd580-b6c177a0-479e85-2183a70a@10.167.131.33
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: kpml, telephone-event
Remote-Party-ID: <sip:0019088873092@10.74.66.5>;party=called;screen=no;privacy=off
Contact: <sip:90019088873090@10.74.66.5:5060;transport=tcp>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.5.3.S6
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 335

v=0
o=CiscoSystemsSIP-GW-UserAgent 720 2571 IN IP4 10.74.66.5
s=SIP Call
c=IN IP4 10.74.66.5
t=0 0
m=audio 9088 RTP/AVP 0 18 100 101
c=IN IP4 10.74.66.5
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Aug  9 17:19:32.853: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.167.131.33:5060;branch=z9hG4bK6d35f92b76b51d
From: <sip:20382@10.167.131.33>;tag=57197010~a6b8e9be-4836-49e6-b65d-1c212bf73946-117107293
To: <sip:90019088873090@10.74.66.5>;tag=9B3BBDC-1E57
Date: Thu, 09 Aug 2018 18:19:28 CST
Call-ID: 5d9cd580-b6c177a0-479e85-2183a70a@10.167.131.33
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: kpml, telephone-event
Remote-Party-ID: <sip:0019088873092@10.74.66.5>;party=called;screen=no;privacy=off
Contact: <sip:90019088873090@10.74.66.5:5060;transport=tcp>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.5.3.S6
Require: timer
Session-Expires:  1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 335

v=0
o=CiscoSystemsSIP-GW-UserAgent 720 2571 IN IP4 10.74.66.5
s=SIP Call
c=IN IP4 10.74.66.5
t=0 0
m=audio 9088 RTP/AVP 0 18 100 101
c=IN IP4 10.74.66.5
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Aug  9 17:19:32.970: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:90019088873090@10.74.66.5:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.167.131.33:5060;branch=z9hG4bK6d36104563566f
From: <sip:20382@10.167.131.33>;tag=57197010~a6b8e9be-4836-49e6-b65d-1c212bf73946-117107293
To: <sip:90019088873090@10.74.66.5>;tag=9B3BBDC-1E57
Date: Thu, 09 Aug 2018 17:19:28 GMT
Call-ID: 5d9cd580-b6c177a0-479e85-2183a70a@10.167.131.33
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 249

v=0
o=CiscoSystemsCCM-SIP 57197010 1 IN IP4 10.167.131.33
s=SIP Call
c=IN IP4 10.86.102.47
b=TIAS:64000
b=CT:64
b=AS:64
t=0 0
m=audio 24670 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Aug  9 17:19:59.584: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:90019088873090@10.74.66.5:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.167.131.33:5060;branch=z9hG4bK6d363b5f3adf0a
From: <sip:20382@10.167.131.33>;tag=57197010~a6b8e9be-4836-49e6-b65d-1c212bf73946-117107293
To: <sip:90019088873090@10.74.66.5>;tag=9B3BBDC-1E57
Date: Thu, 09 Aug 2018 17:19:28 GMT
Call-ID: 5d9cd580-b6c177a0-479e85-2183a70a@10.167.131.33
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
P-Asserted-Identity: <sip:20382@10.167.131.33>
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0


Aug  9 17:19:59.614: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.167.131.33:5060;branch=z9hG4bK6d363b5f3adf0a
From: <sip:20382@10.167.131.33>;tag=57197010~a6b8e9be-4836-49e6-b65d-1c212bf73946-117107293
To: <sip:90019088873090@10.74.66.5>;tag=9B3BBDC-1E57
Date: Thu, 09 Aug 2018 18:19:59 CST
Call-ID: 5d9cd580-b6c177a0-479e85-2183a70a@10.167.131.33
Server: Cisco-SIPGateway/IOS-15.5.3.S6
CSeq: 102 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=1329,OS=265800,PR=1261,OR=252200,PL=0,JI=3,LA=0,DU=26
Content-Length: 0

 

Labels:
0 Helpful
2 Replies 2

Jaime Valencia
Cisco Employee
Cisco Employee

‎08-09-2018 01:03 PM

You need to check the whole dial plan to find out what you could be matching, and not just the RPs/TPs.

If all the possible RPs and TPs have the secondary dial tone checked, then that means something else is being matched.

Use DNA or route plan report.

HTH

java

if this helps, please rate
0 Helpful

stanzin
Level 1
Level 1
In response to Jaime Valencia

‎08-10-2018 07:49 AM

Jaime, thanks for your response
Yes, I checked the call routing and it looked good on my side
I did added "voice call send-alert" and the below command ( under the pots dial peer) which was missing from my config it still didn't fixed my issue, but it did reduce the wait time from 15 sec to 7 sec.
progress_ind alert enable 8
progress_ind connect enable 8
progress_ind progress enable 8

Any suggestions ?
0 Helpful
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