Contents

Introduction

Problem: Caller Hears Fast Busy Tone when Calls from IP Phone 4002 to 4001

Solution

Step 1. Cisco Unified Communication Manager (CUCM) Log Analysis.

Step 2. CUBE SP Log Analysis.

Step 3. Base on Troubleshoot Step 1 and 2 Confirm it Hit Bug.

Workaround

New Issue after you Apply the Workaround

Use SIP Header-Editor to Fix this Issue

Procedure

Verify

No Diversion Header Modification

SIP Hearder-Editor Match Diversion Header

Introduction

This document describes how to troubleshoot Cisco Unified Border Element (SP Edition) (CUBE SP) when it rejects the internal call, which is config fowarded to PSTN number.

Call Flow:  Internal IP phone 4002  calls internal IP phone 4001,  all calls on ip phone 4001 are foward to a configured PSTN number.

Problem: Caller Hears Fast Busy Tone when Calls from IP Phone 4002 to 4001

Caller uses IP Phone 1  to call another IP Phone 2, the IP Phone 2 is configured to forwad all calls to an exteral PSTN number . the call failed to connect the PSTN Phone, PSTN phone does not ring and the caller hears fast busy tone.

Solution

These are the steps to troubleshoot the issue.

Step 1. Cisco Unified Communication Manager (CUCM) Log Analysis.

From CUCM logs, can see error message coming from CUBE  SP.

SIP/2.0 604 Does Not Exist Anywhere

Detail Message:

SIP/2.0 604 Does Not Exist Anywhere from cube SP

82645958.001 |13:08:46.297 |AppInfo  |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.4.15.253 on port 5060 index 18491

[19580587,NET]

INVITE sip:+612xxxxxxxx@10.x.x.x:5060 SIP/2.0

Via: SIP/2.0/TCP 10.4.15.5:5060;branch=z9hG4bK3cc7264a831cc4

From: <sip:+612xxxxxxx@10.x.x.x>;tag=8162255~9cbf8c07-9c9b-758f-e658-bebd74e53d96-40280558

To: <sip:+614xxxxxxx@10.4.15.253>

Date: Fri, 17 Nov 2017 02:08:46 GMT

Call-ID: 3cf99080-a0e144ae-3692cb-50f040a@10.x.x.x

Supported: 100rel,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

Supported: Geolocation

Call-Info: <sip:10.x.x.x:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED

Cisco-Guid: 1022988416-0000065536-0000118822-0084870154

Session-Expires:  1800

Diversion: <sip:9180@10.x.x.x>;reason=unconditional;privacy=off;screen=yes

P-Asserted-Identity: x <sip:+612xxxxxxxx@10.x.x.x>

Remote-Party-ID: x <sip:+612xxxxxxxxx@10.x.x.x>;party=calling;screen=yes;privacy=off

Step 2. CUBE SP Log Analysis.

From CUBE SP logs, you can see that the call did not pass source number analsysis, as it does not match any entry.

inside   na-src-prefix-table

Diversion: <sip:9180@10.x.x.x>;reason=unknown;privacy=off;screen=yes

Routing fails.

SBC Index = 0X00000001

Config set Index = 0X0000270F

Source Account = CUCM-TL1

Source Adjacency = CUCM-cust01-1

Calling Address Type = 0X00030000

Called Address Type = 0X00030000

Calling Address = 9180

Called Address = +614xxxxxxxx

Step 3. Base on Troubleshoot Step 1 and 2 Confirm it Hit Bug.

This hits the konwn bug  CSCup67940

CUCM needs to send E.164 number in diversion header for extend&connect.

https://bst.cloudapps.cisco.com/bugsearch/bug/CSCup67940/?referring_site=bugquickviewredir

Workaround:

Unless we make modification in the CUBE to accept the invite from diversion header contains phone DN such as 26708 <sip:26708@58.162.59.181>;reason=unknown;privacy=off;screen=yes

Workaround

According to the workaround, allow the number in Diversion header.

This can be done to add a new entry in this  na-src-prefix-table.

na-src-prefix-table   xxxxx

entry  10

action accept

match-prefix  9

New Issue after you Apply the Workaround

After you apply this workaroud, the call is successfully connected but a five digit extension number is sent to the Service Provider.

Use SIP Header-Editor to Fix this Issue

Tested in the lab, as you use  SIP header-editor to modify Diversion header in CUBE SP, it connects the call successfully and sends e164 number to service provider.

Procedure

In the lab testing,  IP Phone 4002  calls  4001, on IP phone 4001 call foward all to 60006009 ( PSTN ) number.

sip header-editor donnietest

store-rule entry 1

condition header-name Diversion header-value regex-match "sip:4\(...\)" store-as diversionuri

header diversion entry 1

action replace-value value "<sip:+888888884${diversionuri}@10.66.75.51>;reason=unconditional;

privacy=off;screen=yes"

condition header-name Diversion header-value regex-match "sip:4\(...\)@"

adjacency sip donniecucm

editor-type editor

header-editor inbound donnietest

Verify

No Diversion Header Modification

Without any Diversion Header modification, you can see the invite from CUCM the Diversion Header is  below

Diversion: <sip:4001@10.66.75.51>;reason=unconditional;privacy=off;screen=yes

INVITE sip:60006099@10.66.75.33:5068 SIP/2.0

Via: SIP/2.0/TCP 10.66.75.51:5060;branch=z9hG4bK1ef607cac8bd6

From: "agent2-4002" <sip:4002@10.66.75.51>;tag=194346~4c742393-721f-476b-82c3-bc13f8a9c6cd-22765770

To: <sip:60006099@10.66.75.33>

Date: Sun, 19 Nov 2017 23:39:16 GMT

Call-ID: d9ad6f80-a1211624-1eee8-334b420a@10.66.75.51

Supported: timer,resource-priority,replaces

Min-SE:  1800

User-Agent: Cisco-CUCM11.5

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

Supported: Geolocation

Call-Info: <sip:10.66.75.51:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP

Session-ID: 223cb8ec818c0c0dd669d19baa194344;remote=00000000000000000000000000000000

Cisco-Guid: 3652022144-0000065536-0000000027-0860570122

Session-Expires:  1800

Diversion: <sip:4001@10.66.75.51>;reason=unconditional;privacy=off;screen=yes

P-Asserted-Identity: "agent2-4002" <sip:4002@10.66.75.51>

Remote-Party-ID: "agent2-4002" <sip:4002@10.66.75.51>;party=calling;screen=yes;privacy=off

Contact: <sip:4002@10.66.75.51:5060;transport=tcp>

Max-Forwards: 69

Content-Length: 0

SIP Hearder-Editor Match Diversion Header

SIP header-editor match  the Diversion Header Start with  sip:4xxx@ , then make it +E164 format

It can be seen after sip header-editor. In Diversion Header, 4001  has been modified to +888888884001

Diversion: <sip:+888888884001@10.66.75.51>;reason=unconditional;privacy=off;screen=yes

MSG-6401-0027-69FECA-0747 at 01:48:38, 20 November 2017 (491542613 ms):  0X01000E2059EBD60A

A module has returned a message after editing.

Editor name                    = donnietest

Editor config set              = 0X00000000

This is the message after you edit.

INVITE sip:60006099@10.66.75.33:5068 SIP/2.0

Supported: X-cisco-srtp-fallback

Via: SIP/2.0/TCP 10.66.75.51:5060;branch=z9hG4bK1f11c18671c97

From: "agent2-4002" <sip:4002@10.66.75.51>;tag=194931~4c742393-721f-476b-82c3-bc13f8a9c6cd-22765859

To: <sip:60006099@10.66.75.33>

Date: Mon, 20 Nov 2017 02:13:12 GMT

Call-ID: 5ac33180-a1213a38-1f045-334b420a@10.66.75.51

Min-SE: 1800

User-Agent: Cisco-CUCM11.5

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

CSeq: 101 INVITE

Expires: 180

Allow-Events: presence, kpml

Call-Info: <sip:10.66.75.51:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP

Session-ID: 223cb8ec818c0c0dd669d19baa194929;remote=00000000000000000000000000000000

Cisco-Guid: 1522741632-0000065536-0000000050-0860570122

Session-Expires: 1800

Diversion: <sip:+888888884001@10.66.75.51>;reason=unconditional;privacy=off;screen=yes

P-Asserted-Identity: "agent2-4002" <sip:4002@10.66.75.51>

Remote-Party-ID: "agent2-4002" <sip:4002@10.66.75.51>;party=calling;screen=yes;privacy=off

Contact: <sip:4002@10.66.75.51:5060;transport=tcp>

Max-Forwards: 69

Content-Length: 0

MSG-6401-0028-69FECA-0885 at 01:48:38, 20 November 2017 (491542613 ms):  0X01000E2059EBD60A

The edits are made on the message.

This is the message after you edit

INVITE sip:60006099@10.66.75.33:5068 SIP/2.0

Supported: X-cisco-srtp-fallback

Via: SIP/2.0/TCP 10.66.75.51:5060;branch=z9hG4bK1f11c18671c97

From: "agent2-4002" <sip:4002@10.66.75.51>;tag=194931~4c742393-721f-476b-82c3-bc13f8a9c6cd-22765859

To: <sip:60006099@10.66.75.33>

Date: Mon, 20 Nov 2017 02:13:12 GMT

Call-ID: 5ac33180-a1213a38-1f045-334b420a@10.66.75.51

Min-SE: 1800

User-Agent: Cisco-CUCM11.5

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

CSeq: 101 INVITE

Expires: 180

Allow-Events: presence, kpml

Call-Info: <sip:10.66.75.51:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP

Session-ID: 223cb8ec818c0c0dd669d19baa194929;remote=00000000000000000000000000000000

Cisco-Guid: 1522741632-0000065536-0000000050-0860570122

Session-Expires: 1800

Diversion: <sip:+888888884001@10.66.75.51>;reason=unconditional;privacy=off;screen=yes

P-Asserted-Identity: "agent2-4002" <sip:4002@10.66.75.51>

Remote-Party-ID: "agent2-4002" <sip:4002@10.66.75.51>;party=calling;screen=yes;privacy=off

Contact: <sip:4002@10.66.75.51:5060;transport=tcp>

Max-Forwards: 69

Content-Length: 0