Re: Busy Tone when calling Outbound

abuahmed.sd · ‎01-03-2022

Hello, all of there,

When I place a call from any extension in our cucm some times is the call established and sometimes gives me a busy tone at random times, Knowing that I can make more than 300 calls simultaneously inbound or outbound based on my configuration. and it was working fine before.
can someone explain this issue?

note:

+ I use CUCM v 11.5

+ and I have more than 900 extensions of multiple IP phone models.

+ just 700 can have permission to call outbound.

+ the CDR report mentions that no more than 90 calls per day.

+ when the call is established it has s distortion with it.

Nithin Eluvathingal · ‎01-03-2022

What PSTN connectivity you have ? Did you check if the calls are landing on gateway ?

abuahmed.sd · ‎01-04-2022

I have an optical fiber cable to connect from PSTN. and the signal of the fiber cable is -23.6dB.

I forget that: when the call is established it has s distortion with it.

and How can I check they are landing on the gateway?

Nithin Eluvathingal · ‎01-05-2022

Check with ISP if its SIP or E1/T1.

You might be using a Cisco Router to connect the PSTN line. Check the router configuration and we need to debug the gateway.

abuahmed.sd · ‎01-05-2022

Thanks, @Nithin Eluvathingal for your reply.

It is a SIP line from PSTN

and the failure call details are shown below:

the Calling number: 1217

the called number: 90506069610

49259081.002 |10:45:08.268 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.90.58.110 on port 51517 index 675556 with 1520 bytes:
[781782264,NET]
INVITE sip:90506069610@192.168.240.12;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.90.58.110:51517;branch=z9hG4bK0f9df516
From: "Musab Abdelkareem" <sip:1217@192.168.240.12>;tag=84802d76bb77ac581b5b0366-19f9d38e
To: <sip:90506069610@192.168.240.12>
Call-ID: 84802d76-bb77028d-3fe50fab-52ee4412@10.90.58.110
Max-Forwards: 70
Date: Tue, 04 Jan 2022 07:45:06 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP9971/9.4.2
Contact: <sip:44d53073-03a6-ed2a-00d8-1424ad79db2e@10.90.58.110:51517;transport=tcp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Musab Abdelkareem" <sip:1217@192.168.240.12>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.0.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 352
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 24439 0 IN IP4 10.90.58.110
s=SIP Call
t=0 0
m=audio 25116 RTP/AVP 102 0 8 116 18 101
c=IN IP4 10.90.58.110
a=rtpmap:102 L16/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Timestamp: 3724137908268
UTC Timestamp:3724127108268
Source Filename: SDL001_100_001321.txt.gz

==================
49259084.001 |10:45:08.269 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.90.58.110 on port 51517 index 675556
[781782265,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.90.58.110:51517;branch=z9hG4bK0f9df516
From: "Musab Abdelkareem" <sip:1217@192.168.240.12>;tag=84802d76bb77ac581b5b0366-19f9d38e
To: <sip:90506069610@192.168.240.12>
Date: Tue, 04 Jan 2022 07:45:08 GMT
Call-ID: 84802d76-bb77028d-3fe50fab-52ee4412@10.90.58.110
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0

Timestamp: 3724137908269
UTC Timestamp:3724127108269
Source Filename: SDL001_100_001321.txt.gz

=========================

49259158.001 |10:45:08.279 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.90.58.110 on port 51517 index 675556
[781782266,NET]
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP 10.90.58.110:51517;branch=z9hG4bK0f9df516
From: "Musab Abdelkareem" <sip:1217@192.168.240.12>;tag=84802d76bb77ac581b5b0366-19f9d38e
To: <sip:90506069610@192.168.240.12>;tag=261844280~daac9a36-990f-40f6-acf2-f6fd9bc0b135-23454874
Date: Tue, 04 Jan 2022 07:45:08 GMT
Call-ID: 84802d76-bb77028d-3fe50fab-52ee4412@10.90.58.110
CSeq: 101 INVITE
Allow-Events: presence
Server: Cisco-CUCM11.5
Session-ID: 00000000000000000000000000000000;remote=4769f33079398ba38a271aa261844280
Reason: Q.850; cause=41
Remote-Party-ID: <sip:90506069610@192.168.240.12;user=phone>;party=x-cisco-original-called;privacy=off
Content-Length: 0

Timestamp: 3724137908279
UTC Timestamp:3724127108279
Source Filename: SDL001_100_001321.txt.gz

====================

49259159.002 |10:45:08.284 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.90.58.110 on port 51517 index 675556 with 454 bytes:
[781782267,NET]
ACK sip:90506069610@192.168.240.12;user=phone SIP/2.0
Via: SIP/2.0/TCP 10.90.58.110:51517;branch=z9hG4bK0f9df516
From: "Musab Abdelkareem" <sip:1217@192.168.240.12>;tag=84802d76bb77ac581b5b0366-19f9d38e
To: <sip:90506069610@192.168.240.12>;tag=261844280~daac9a36-990f-40f6-acf2-f6fd9bc0b135-23454874
Call-ID: 84802d76-bb77028d-3fe50fab-52ee4412@10.90.58.110
Max-Forwards: 70
Date: Tue, 04 Jan 2022 07:45:06 GMT
CSeq: 101 ACK
Content-Length: 0

Timestamp: 3724137908284
UTC Timestamp:3724127108284
Source Filename: SDL001_100_001321.txt.gz

--------------------------------------

this debugs with TranslatorX

abuahmed.sd · ‎01-09-2022

any news on this problem?

Roger Kallberg · ‎01-09-2022

Please post your running configuration and the output from debug ccsip message and debug voip ccapi inout when active at the same time on the gateway and make an outbound call. Collect the information in text files, one with the configuration and another with the debug output, and attach them to your post.

abuahmed.sd · ‎01-03-2022

are you think that from our connectivity method?