Re: UC560 calling number (ANI) in transfered call

jfrodriguez23 · ‎06-11-2018

Hello,

My name is Julián.

We have a customer with a UC560 Pbx. We have a problem with transfered calls. One call is received in UC560, and this call is trasfered to other system by one voip dial-peer.

Calling Number A (external number - PRI E1 port)

Called Number B (ip phone registered in UC560)

Transfered number (external device)

This option works fine, the ip phone (B) can to transfer the call from A to C, but:

This other system needs to get several fields from the call, for example who is the originating calling number (number A). I have added the comand clid network-provided in dial peer but this option does not works for me.

Is it posible to show in number C, the original calling number (number A)?

Could you help me?

Kind Regards,

Daniele Giordano · ‎06-13-2018

Can you post the configuration of the UC?

Do you have the option 'call-forward system redirecting-expanded' under telephony-service menu?

Regards.

jfrodriguez23 · ‎06-13-2018

Hello Daniele,

i have attached sh_run.

I do not have call-forward system redirecting-expanded' under telephony-service menu.

In SIP debug, this is my first invite to trasnsfered extension:

Sent:
INVITE sip:600@192.168.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5D1C1E
Remote-Party-ID: "Ingrid/Eva" <sip:200@192.168.0.1>;party=calling;screen=no;privacy=off
From: "Ingrid/Eva" <sip:200@192.168.0.1>;tag=261690-90F
To: <sip:600@192.168.0.2>
Date: Wed, 13 Jun 2018 13:22:16 GMT
Call-ID: A077BFA8-6E4311E8-81CCEA7B-A85B6B69@192.168.0.1
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2683842855-1849889256-2177297019-2824563561
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1528896136
Contact: <sip:200@192.168.0.1:5060>
Call-Info: <sip:192.168.0.1:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 267

v=0
o=CiscoSystemsSIP-GW-UserAgent 124 4656 IN IP4 192.168.0.1
s=SIP Call
c=IN IP4 192.168.0.1
t=0 0
m=audio 19530 RTP/AVP 8 101 19
c=IN IP4 192.168.0.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

screen=no, ¿could be do this the problem?

Regards,

Daniele Giordano · ‎06-13-2018

Hi, try to remove the command

calling-number local

from your telephony-service

This command is used to replace the original calling-party with the forwarding-party number.

Regards.

jfrodriguez23 · ‎06-15-2018

Hello Daniele, I was some testa yesterday, and this removed command does not work. We have the same problem ....

Daniele Giordano · ‎06-15-2018

add a new debug ccsip, please

regards

jfrodriguez23 · ‎06-15-2018

Hello,

Calling number 911939521
Called number 932128108 (internal extension 200)
The called number is an AA

Hour of call: 10:54

The call was attended and transfered at 11:00

Transfered extension: 600

I hace attached ccsip all and q931 debugs.

Regards

Daniele Giordano · ‎06-15-2018

In the trace I cannot see calls to extension 600.

I see a call to extension 299:

INVITE sip:299@10.1.10.1:5060 SIP/2.0

Via: SIP/2.0/UDP 10.1.10.2:5060;x-route-tag="tgrp:ALL_T1E1";branch=z9hG4bK11F7548

Remote-Party-ID: <sip:911939521@10.1.10.2>;party=calling;screen=yes;privacy=off

From: <sip:911939521@10.1.10.2>;tag=97D1F98-2427

To: <sip:299@10.1.10.1>

...

jfrodriguez23 · ‎06-15-2018

It is this (appear in logs file),

112887: //12521/72E4AE5A8C43/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:600@192.168.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK12641875
Remote-Party-ID: "Ingrid/Eva" <sip:200@192.168.0.1>;party=calling;screen=no;privacy=off
From: "Ingrid/Eva" <sip:200@192.168.0.1>;tag=9838048-2463
To: <sip:600@192.168.0.2>
Date: Fri, 15 Jun 2018 09:00:57 GMT
Call-ID: 73A7FEEE-6FB111E8-8C49EA7B-A85B6B69@192.168.0.1
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1927589466-1873875432-2353261179-2824563561
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1529053257
Contact: <sip:200@192.168.0.1:5060>
Call-Info: <sip:192.168.0.1:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 268

Daniele Giordano · ‎06-15-2018

I've better checked the logs.

I see that new call to extension 600 uses dial-peer [4000]:

dial-peer voice 4000 voip
destination-pattern 6..
b2bua
session protocol sipv2
session target ipv4:192.168.0.2
dtmf-relay sip-notify rtp-nte
codec g711alaw
clid network-provided

Can you try to rmeove from this dp the command 'clid network-provided'?

What is the result?

And what happens if you use the command 'clid override rdnis'?

Regards.

jfrodriguez23 · ‎06-18-2018

Hello Daniele, I removed clid network-provided, and now i see an update of connected number. The provider is checking if this solutin is valid. I will update as soon as posible,

Regards,

Daniele Giordano · ‎06-19-2018

ok glad to hear it, let me know the feedback of the provider

regards

jfrodriguez23 · ‎07-03-2018

Hello Danielle,

I am sorry by delay.

With removed command clid network-provided:

Calling number: 913346155

Called number: 932128108

This call is attended by ip phone (ext. 200)

Ext. 200 transfer to 600 by dial peer 4000. In 600, the calling number is 200.

I have checked what the clid override rdnis command it is not available in voice dial peer. It is supported in pots dial peer only.