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CUCM 11.5 CFA - Internal/External issue

Neil Sheridan
Level 1
Level 1

‎07-09-2017 08:27 PM - edited ‎03-17-2019 10:44 AM

Hi,

I have a new installation of CUCM/CUC 11.5 with a SIP trunk to ITSP via CUBE. The system is working with inbound/outbound calls for our pilot. We have a range of 25 numbers for the pilot.

I am trying to configure Call Forwarding, either CFNA external or CFA to external. Detailed below are some of the current configurations:

Service Parameter for CFA is "With Activating Device/Line CSS"

Route patterns for external numbers are all configured with "0. for external line" and "discard predot"

Manually forwarding (transfer) the call works perfectly.

Our ITSP requires each number/extension to be authenticated via CUBE config.

Below is the current call flow:

IP Phone -> CUCM > CUBE -> SIP Trunk -> ITSP SIP Trunk -> PSTN

Now, If I set CFA from my line (234) to my mobile and place a call to me extension I receive a busy tone. If an internal user calls my extension they receive the "Your call could not be completed s dialled" message.

I've looked through the debug logs attached and I see some information that I believe relates to the ITSP no accepting the Calling number being presented. The ITSP will NOT accept and number other than the 25 provide with an area code prefix (0862702734). From what I can see the CFA is sending the originating calls number. I've tried a few permutations of the CALLING number to no avail.

CUCM = 10.1.1.135

CUBE = 10.1.1.7

CUBE - ITSP INT = 172.22.127.90

CUBE GW = 172.22.127.89

CUBE DNS = 202.147.129.26 and 30

ITSP Realm - amcomvoice.ipsystems.com.au

Translation rules / Profiles are:

voice class sip-profiles 1

request INVITE sip-header Privacy add "Privacy: id"

voice translation-rule 1

rule 1 /^08627027/ /2/

voice translation-profile Extensions

translate called 1

The dial-peers on CUBE are:

dial-peer voice 10 voip

description Inbound to/from CUCM

huntstop

destination-pattern 08627027..

session protocol sipv2

session target ipv4:10.1.1.135

incoming called-number .

voice-class codec 1 

dtmf-relay rtp-nte

dial-peer voice 20 voip

description To-From SIP Proxy

translation-profile incoming Extensions

huntstop

destination-pattern .T

session protocol sipv2

session target sip-server

incoming called-number 08627027..

voice-class codec 1 

voice-class sip profiles 1

dtmf-relay rtp-nte

dial-peer voice 11 voip

description inbound to/from CUCM

huntstop

destination-pattern 2..

session protocol sipv2

session target ipv4:10.1.1.135

incoming called-number .

voice-class codec 1 

dtmf-relay rtp-nte

Unfortunately this is now beyond my CCNAV, and I would very much welcome your assistance.

Regards

N

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4 Replies 4

james.buchanan1
Level 1
Level 1

‎07-09-2017 11:09 PM

Hello,

You need to set the Diversion header to a number that is included in your range. You do this by creating a SIP profile, and by applying that SIP profile to your outbound SIP dial peer. Here's an example:

voice class sip-profiles 1

request INVITE sip-header Diversion modify "<sip:1234@(.*)>" "<sip:+12025555555@192.168.0.1>"

In this example, "1234" is the extension that diverted the call. "+12025555555" is a valid number the ITSP is expecting. 192.168.0.1 is the IP address to which the call is being directed, in this case the ITSP-facing interface. in your case, it would be "10.1.1.135"

So, based on your example your header should look something like this:

request INVITE sip-header Diversion modify "<sip:(.*)@(.*)>" "<sip:+12025555555@10.1.1.135>"

It does not matter which number you use in the modified diversion header as long as it is on your SIP trunk.

On your outbound dial peer, you would set voice-class sip-profile 1 to apply this.

Thanks,

James

0 Helpful

Neil Sheridan
Level 1
Level 1
In response to james.buchanan1

‎07-10-2017 01:03 AM

Hi James,

Thank you for your reply. I did come to the conclusion it was going to be something like that. I've made the changes suggested, but unfortunately the call is still not transferred. I performed some debugs to determine which OUTGOING DP required the profile added to, it "seems" DP 20, however no change. Below is the CUBE config and some of the debug log after applying.

!

voice class sip-profiles 1

request INVITE sip-header Diversion modify "<sip:(.*)@(.*)>" "<sip:0862702734@10.1.1.135>"

!

dial-peer voice 20 voip

description To-From SIP Proxy

translation-profile incoming Extensions

huntstop

destination-pattern .T

session protocol sipv2

session target sip-server

incoming called-number 08627027..

voice-class codec 1 

voice-class sip profiles 1

dtmf-relay rtp-nte

!

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=20

Jul 10 07:55:39.755: //-1/28313E800000/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=0450633094, Peer Info Type=DIALPEER_INFO_SPEECH

Jul 10 07:55:39.755: //-1/28313E800000/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=0450633094

Jul 10 07:55:39.755: //-1/28313E800000/DPM/dpMatchCore:

   Dial String=0450633094, Expanded String=0450633094, Calling Number=

   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH

Jul 10 07:55:39.755: //-1/28313E800000/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=20 Is Matched (0 digits)

Jul 10 07:55:39.755: //-1/28313E800000/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

Jul 10 07:55:39.755: //-1/28313E800000/DPM/dpMatchSafModulePlugin:

   dialstring=0450633094, saf_enabled=0, saf_dndb_lookup=1, dp_result=0

Jul 10 07:55:39.755: //-1/28313E800000/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=20

Jul 10 07:55:39.755: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=0450633094, Called Number=0450633094, Peer Info Type=DIALPEER_INFO_SPEECH

Jul 10 07:55:39.755: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=0450633094

Jul 10 07:55:39.755: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:

   Dial String=0450633094, Expanded String=0450633094, Calling Number=0450633094T

   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH

Jul 10 07:55:39.759: //-1/28313E800000/DPM/dpMatchSafModulePlugin:

   dialstring=0450633094, saf_enabled=1, saf_dndb_lookup=1, dp_result=0

Jul 10 07:55:39.759: //-1/28313E800000/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=20

Thank you for taking the time to help.

n

0 Helpful

Neil Sheridan
Level 1
Level 1
In response to Neil Sheridan

‎07-10-2017 06:47 PM

I've run another debug and can  now see the Diversion occuring, however it's complaining about no matching outgoing dial peer as per below :(

Jul 11 01:45:13.000: //26/92E089800000/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:0450633094@amcomvoice.ipsystems.com.au:5060 SIP/2.0

Via: SIP/2.0/UDP 172.22.127.90:5060;branch=z9hG4bK1B1E58

From: <sip:0862431267@amcomvoice.ipsystems.com.au>;tag=108564-7CA

To: <sip:0450633094@amcomvoice.ipsystems.com.au>

Date: Tue, 11 Jul 2017 01:45:12 GMT

Call-ID: 6A4623BE-651111E7-803FE95F-D3292FAF@amcomvoice.ipsystems.com.au

Supported: 100rel,timer,resource-priority,replaces,histinfo,sdp-anat

Min-SE:  1800

Cisco-Guid: 2464188800-0000065536-0000000448-2264989962

User-Agent: Cisco-SIPGateway/IOS-15.6.3.M2

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

CSeq: 101 INVITE

Timestamp: 1499737512

Contact: <sip:0862431267@172.22.127.90:5060>

History-Info: <sip:234@172.22.127.90?Reason=sip%3Bcause%3D302%3Btext%3D%22unconditional%22>;index=1,<sip:0450633094@amcomvoice.ipsystems.com.au:5060>;index=1.1

Expires: 300

Allow-Events: telephone-event

Max-Forwards: 26

Diversion: "Neil Sheridan"<sip:0862702734@amcomvoice.ipsystems.com.au>;privacy=off;reason=unconditional;screen=yes

P-Asserted-Identity: <sip:0862431267@amcomvoice.ipsystems.com.au>

Session-ID: ebf8bdc974405bbf9f0f905c38218d05;remote=00000000000000000000000000000000

Session-Expires:  1800

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 345

Privacy: id

v=0

o=CiscoSystemsSIP-GW-UserAgent 25 3343 IN IP4 172.22.127.90

s=SIP Call

c=IN IP4 172.22.127.90

t=0 0

m=audio 16414 RTP/AVP 0 8 100 101 19

c=IN IP4 172.22.127.90

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:100 X-NSE/8000

a=fmtp:100 192-194

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:19 CN/8000

a=ptime:20

Jul 11 01:45:13.596: //26/92E089800000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.22.127.90:5060;branch=z9hG4bK1B1E58

From: <sip:0862431267@amcomvoice.ipsystems.com.au>;tag=108564-7CA

To: <sip:0450633094@amcomvoice.ipsystems.com.au>

Call-ID: 6A4623BE-651111E7-803FE95F-D3292FAF@amcomvoice.ipsystems.com.au

CSeq: 101 INVITE

Timestamp: 1499737512

Jul 11 01:45:13.900: //26/92E089800000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 604 Does not exist anywhere

Via: SIP/2.0/UDP 172.22.127.90:5060;branch=z9hG4bK1B1E58

From: <sip:0862431267@amcomvoice.ipsystems.com.au>;tag=108564-7CA

To: <sip:0450633094@amcomvoice.ipsystems.com.au>;tag=SDd0n0599-1035143415-1499737513042

Call-ID: 6A4623BE-651111E7-803FE95F-D3292FAF@amcomvoice.ipsystems.com.au

CSeq: 101 INVITE

Timestamp: 1499737512

Content-Length: 0

Jul 11 01:45:13.904: //26/92E089800000/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:0450633094@amcomvoice.ipsystems.com.au:5060 SIP/2.0

Via: SIP/2.0/UDP 172.22.127.90:5060;branch=z9hG4bK1B1E58

From: <sip:0862431267@amcomvoice.ipsystems.com.au>;tag=108564-7CA

To: <sip:0450633094@amcomvoice.ipsystems.com.au>;tag=SDd0n0599-1035143415-1499737513042

Date: Tue, 11 Jul 2017 01:45:12 GMT

Call-ID: 6A4623BE-651111E7-803FE95F-D3292FAF@amcomvoice.ipsystems.com.au

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Session-ID: ;remote=

Content-Length: 0

Jul 11 01:45:13.908: //26/92E089800000/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 604 Does not exist anywhere

Via: SIP/2.0/UDP 172.22.127.90:5060;branch=z9hG4bK1B1E58

From: <sip:0862431267@amcomvoice.ipsystems.com.au>;tag=108564-7CA

To: <sip:0450633094@amcomvoice.ipsystems.com.au>;tag=SDd0n0599-1035143415-1499737513042

Call-ID: 6A4623BE-651111E7-803FE95F-D3292FAF@amcomvoice.ipsystems.com.au

CSeq: 101 INVITE

Timestamp: 1499737512

Content-Length: 0

Jul 11 01:45:13.908: //26/92E089800000/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:0450633094@amcomvoice.ipsystems.com.au:5060 SIP/2.0

Via: SIP/2.0/UDP 172.22.127.90:5060;branch=z9hG4bK1B1E58

From: <sip:0862431267@amcomvoice.ipsystems.com.au>;tag=108564-7CA

To: <sip:0450633094@amcomvoice.ipsystems.com.au>;tag=SDd0n0599-1035143415-1499737513042

Date: Tue, 11 Jul 2017 01:45:12 GMT

Call-ID: 6A4623BE-651111E7-803FE95F-D3292FAF@amcomvoice.ipsystems.com.au

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Session-ID: ;remote=

Content-Length: 0

Jul 11 01:45:13.908: //25/92E089800000/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/TCP 10.1.1.135:5060;branch=z9hG4bK3592d170531

From: "Neil Sheridan" <sip:234@10.1.1.135>;tag=80243~7b3fd6e6-1684-48fe-880c-6543208d27e6-24846300

To: <sip:0450633094@10.1.1.17>;tag=1088F4-F83

Date: Tue, 11 Jul 2017 01:45:12 GMT

Call-ID: 92e08980-96412da8-27d-8701010a@10.1.1.135

CSeq: 101 INVITE

Allow-Events: telephone-event

Warning: 399 10.1.1.17 "No matching outgoing dial-peer"

Server: Cisco-SIPGateway/IOS-15.6.3.M2

Reason: Q.850;cause=1

Session-ID: 00000000000000000000000000000000;remote=ebf8bdc974405bbf9f0f905c38218d05

Content-Length: 0

0 Helpful

isc@saisd.net
Level 1
Level 1
In response to Neil Sheridan

‎10-09-2019 10:23 AM

Neil,

 

Did you get this problem resolved? If so, can you let us know the solution (I have the seemingly exact same issue).

0 Helpful
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