Re: Problem with call transfer (CME)

mcastillocandia · ‎09-04-2018

Hello

I am currently setting up a CME with a SIP Tunk for calls to a remote destination (external numbers).

Calls between CME and external numbers work OK, but if I attempt to put a call on hold or to transfer it, that call ends abruptly.

In a CM scenario, I would force the SIP Trunk to use MTP, but in CME I don't find a way to force SIP Phones to use MTP (there is a mtp command for ephones though).

Is it possible to force mtp on a SIP Phone in CME?

Is there any other possible cause for the failure of Call on Hold and Call Transfer?

I would appreciate any help with this.

R0g22 · ‎09-04-2018

What codec is negotiated for the call ? What codecs does your ITSP support ? What codec is your MOH file ?
For transfer and forwards, ensure the REFER and 302 is disabled. They are enabled by default on IOS. ITSP's don't want a SIP REFER or 302 and would drop the calls.

voice service voip
sip
no supplementary-service sip-refer
no supplementary-service moved-temp

Just check the commands. There might be a syntax error. :P

mcastillocandia · ‎09-04-2018

Hello
thank you very much for your help

the trace of the call indicates codec negotiation g279

What codec is negotiated for the call ?

g729

What codecs does your ITSP support ?

g729

What codec is your MOH file ?

CSeq: 2 BYE

Reason: Q.850;cause=127;text="interworking unspecified"

CSeq: 2 BYE

Reason: Q.850;cause=16

voice service voip
ip address trusted list
ipv4 192.168.200.2
ipv4 192.168.100.200
ipv4 10.235.146.97
ipv4 0.0.0.0 0.0.0.0
media disable-detailed-stats
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
no supplementary-service sip handle-replaces
supplementary-service media-renegotiate
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
emptycapability
h225 signal overlap
h225 connect-passthru
call start interwork
h245 passthru tcsnonstd-passthru
sip
rel1xx supported "rel1xx"
min-se 90 session-expires 90
registrar server expires max 600 min 60
no update-callerid
early-offer forced
pass-thru content sdp

Trace Call

IPPBX-CorpDesRancagua#

*Sep 4 21:23:57.628: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:962069504@10.235.146.97:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bKCF1EC3

From: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

To: <sip:962069504@10.235.146.97>

Date: Tue, 04 Sep 2018 21:23:57 GMT

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

Supported: rel1xx,timer,resource-priority,replaces,sdp-anat

Min-SE: 90

Cisco-Guid: 2844995241-2948534760-2244510000-0606351652

User-Agent: Cisco-SIPGateway/IOS-15.5.3.S4b

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

CSeq: 101 INVITE

Max-Forwards: 12

Timestamp: 1536096237

Contact: <sip:722605380@192.168.100.200:5060>

Expires: 300

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 278

v=0

o=CiscoSystemsSIP-GW-UserAgent 9216 267 IN IP4 192.168.100.200

s=SIP Call

c=IN IP4 192.168.100.200

t=0 0

m=audio 8230 RTP/AVP 18 101

c=IN IP4 192.168.100.200

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

*Sep 4 21:23:57.669: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bKCF1EC3

From: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

To: <sip:962069504@10.235.146.97>

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

CSeq: 101 INVITE

Timestamp: 1536096237

*Sep 4 21:24:01.436: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bKCF1EC3

From: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

To: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

CSeq: 101 INVITE

Timestamp: 1536096237

Contact: <sip:10.235.146.97:5060;transport=udp>

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

Content-Length: 0

*Sep 4 21:24:02.275: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bKCF1EC3

From: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

To: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

CSeq: 101 INVITE

Timestamp: 1536096237

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

Contact: <sip:10.235.146.97:5060;transport=udp>

Content-Length: 0

*Sep 4 21:24:04.666: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bKCF1EC3

From: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

To: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

CSeq: 101 INVITE

Timestamp: 1536096237

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

Session-Expires: 300;refresher=uac

Require: timer

P-Asserted-Identity: <sip:10.96.63.129>

Contact: <sip:10.235.146.97:5060;transport=udp>

Content-Length: 228

Content-Type: application/sdp

v=0

o=HuaweiUAC3000 2217481 2217481 IN IP4 10.235.146.97

s=Sip Call

c=IN IP4 10.235.146.97

t=0 0

m=audio 32270 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=fmtp:18 annexb=no

*Sep 4 21:24:04.670: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:10.235.146.97:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bKD01D7

From: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

To: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

Date: Tue, 04 Sep 2018 21:23:57 GMT

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

Max-Forwards: 12

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

*Sep 4 21:24:04.755: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:722605380@192.168.100.200:5060 SIP/2.0

Via: SIP/2.0/UDP 10.235.146.97:5060;branch=z9hG4bKan5jtg00ag9lbo2rhpp0sb0000g00.1

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

From: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

To: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

CSeq: 1 INVITE

Contact: <sip:10.235.146.97:5060;transport=udp>

Session-Expires: 300;refresher=uas

Supported: timer

Max-Forwards: 69

Content-Length: 0

*Sep 4 21:24:04.758: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.235.146.97:5060;branch=z9hG4bKan5jtg00ag9lbo2rhpp0sb0000g00.1

From: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

To: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

Date: Tue, 04 Sep 2018 21:24:04 GMT

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

CSeq: 1 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.5.3.S4b

Content-Length: 0

*Sep 4 21:24:04.758: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.235.146.97:5060;branch=z9hG4bKan5jtg00ag9lbo2rhpp0sb0000g00.1

From: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

To: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

Date: Tue, 04 Sep 2018 21:24:04 GMT

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

CSeq: 1 INVITE

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

Allow-Events: telephone-event

Contact: <sip:722605380@192.168.100.200:5060>

Supported: replaces

Supported: sdp-anat

Server: Cisco-SIPGateway/IOS-15.5.3.S4b

Require: timer

Session-Expires: 300;refresher=uas

Supported: timer

Content-Type: application/sdp

Content-Length: 278

v=0

o=CiscoSystemsSIP-GW-UserAgent 9216 267 IN IP4 192.168.100.200

s=SIP Call

c=IN IP4 192.168.100.200

t=0 0

m=audio 8230 RTP/AVP 18 101

c=IN IP4 192.168.100.200

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

*Sep 4 21:24:04.994: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:722605380@192.168.100.200:5060 SIP/2.0

Via: SIP/2.0/UDP 10.235.146.97:5060;branch=z9hG4bKrnsenc00f000h9vo6qj0.1

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

From: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

To: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

CSeq: 1 ACK

Max-Forwards: 69

Content-Length: 228

Content-Type: application/sdp

v=0

o=HuaweiUAC3000 2217481 2217482 IN IP4 10.235.146.97

s=Sip Call

c=IN IP4 10.235.146.97

t=0 0

m=audio 32270 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=fmtp:18 annexb=no

IPPBX-CorpDesRancagua#

call transfer

IPPBX-CorpDesRancagua#

*Sep 4 21:24:13.483: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

OPTIONS sip:192.168.100.200:5060 SIP/2.0

Via: SIP/2.0/UDP 10.235.146.97:5060;branch=z9hG4bKte6uch107oesa8h4k590.1

Call-ID: 89yar78daytf8tavfvcv5rcv6adppfrf@UAC

From: <sip:10.235.146.97:5060>;tag=7avvdacf

To: <sip:192.168.100.200:5060>

CSeq: 1 OPTIONS

Max-Forwards: 69

Content-Length: 0

*Sep 4 21:24:13.484: //1112/B4636E2E85CE/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.235.146.97:5060;branch=z9hG4bKte6uch107oesa8h4k590.1

From: <sip:10.235.146.97:5060>;tag=7avvdacf

To: <sip:192.168.100.200:5060>;tag=252439E-28A

Date: Tue, 04 Sep 2018 21:24:13 GMT

Call-ID: 89yar78daytf8tavfvcv5rcv6adppfrf@UAC

Server: Cisco-SIPGateway/IOS-15.5.3.S4b

CSeq: 1 OPTIONS

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

Allow-Events: telephone-event

Accept: application/sdp

Supported: rel1xx,timer,resource-priority,replaces,sdp-anat

Content-Type: application/sdp

Content-Length: 385

v=0

o=CiscoSystemsSIP-GW-UserAgent 9587 3636 IN IP4 192.168.100.200

s=SIP Call

c=IN IP4 192.168.100.200

t=0 0

m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3

c=IN IP4 192.168.100.200

m=image 0 udptl t38

c=IN IP4 192.168.100.200

a=T38FaxVersion:0

a=T38MaxBitRate:9600

a=T38FaxRateManagement:transferredTCF

a=T38FaxMaxBuffer:200

a=T38FaxMaxDatagram:320

a=T38FaxUdpEC:t38UDPRedundancy

*Sep 4 21:24:13.600: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:10.235.146.97:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bKD1EBC

From: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

To: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

Date: Tue, 04 Sep 2018 21:24:13 GMT

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

Supported: rel1xx,timer,resource-priority,replaces,sdp-anat

Min-SE: 90

Cisco-Guid: 2844995241-2948534760-2244510000-0606351652

User-Agent: Cisco-SIPGateway/IOS-15.5.3.S4b

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

CSeq: 102 INVITE

Max-Forwards: 12

Timestamp: 1536096253

Contact: <sip:722605380@192.168.100.200:5060>

Expires: 300

Allow-Events: telephone-event

Session-Expires: 300;refresher=uac

Content-Type: application/sdp

Content-Length: 290

v=0

o=CiscoSystemsSIP-GW-UserAgent 9216 268 IN IP4 192.168.100.200

s=SIP Call

c=IN IP4 192.168.100.200

t=0 0

m=audio 8230 RTP/AVP 18 101

c=IN IP4 192.168.100.200

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendonly

*Sep 4 21:24:13.722: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:722605380@192.168.100.200:5060 SIP/2.0

Via: SIP/2.0/UDP 10.235.146.97:5060;branch=z9hG4bKan5jtg00ag9lbo2rhpp0sd0000010.1

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

From: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

To: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

CSeq: 2 BYE

Reason: Q.850;cause=127;text="interworking unspecified"

Max-Forwards: 69

Content-Length: 0

*Sep 4 21:24:13.726: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.235.146.97:5060;branch=z9hG4bKan5jtg00ag9lbo2rhpp0sd0000010.1

From: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

To: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

Date: Tue, 04 Sep 2018 21:24:13 GMT

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

Server: Cisco-SIPGateway/IOS-15.5.3.S4b

CSeq: 2 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=67,OS=4020,PR=442,OR=26480,PL=0,JI=0,LA=0,DU=9

Content-Length: 0

*Sep 4 21:24:13.796: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bKD1EBC

From: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

To: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

CSeq: 102 INVITE

Timestamp: 1536096253

Warning: 399 UAC "SS250200F139L990[05537] Request terminated due to Dialog deletion"

Content-Length: 0

*Sep 4 21:24:13.797: //1111/A9932EA985C8/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:10.235.146.97:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bKD1EBC

From: "tes" <sip:722605380@192.168.100.200>;tag=25205AE-183C

To: <sip:962069504@10.235.146.97>;tag=6r5vrpn6-CC-1001

Date: Tue, 04 Sep 2018 21:24:13 GMT

Call-ID: AAEFFF60-AFBF11E8-85CD8130-24243124@192.168.100.200

Max-Forwards: 12

CSeq: 102 ACK

Allow-Events: telephone-event

Content-Length: 0

*Sep 4 21:24:14.590: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

REGISTER sip:192.168.200.2 SIP/2.0

Via: SIP/2.0/UDP 192.168.200.6:5060;branch=z9hG4bK7b2b8a7b

From: <sip:384@192.168.200.2>;tag=f87b204e33a506be27f4eb01-5b8e7e84

To: <sip:384@192.168.200.2>

Call-ID: f87b204e-33a50037-345c47f2-6268ed28@192.168.200.6

Max-Forwards: 70

Session-ID: 1dd6ba2900105000a000f87b204e33a5;remote=00000000000000000000000000000000

Date: Tue, 04 Sep 2018 21:24:13 GMT

CSeq: 1186 REGISTER

User-Agent: Cisco-CP8841/11.7.1

Contact: <sip:18706-42B@192.168.200.6:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-f87b204e33a5>";+u.sip!devicename.ccm.cisco.com="SEPF87B204E33A5";+u.sip!model.ccm.cisco.com="683"

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-8.5.1

Content-Length: 0

Expires: 3600

*Sep 4 21:24:14.591: //1115/B50C588A85D0/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.200.6:5060;branch=z9hG4bK7b2b8a7b

From: <sip:384@192.168.200.2>;tag=f87b204e33a506be27f4eb01-5b8e7e84

To: <sip:384@192.168.200.2>;tag=25247F1-221E

Date: Tue, 04 Sep 2018 21:24:14 GMT

Call-ID: f87b204e-33a50037-345c47f2-6268ed28@192.168.200.6

Server: Cisco-SIPGateway/IOS-15.5.3.S4b

CSeq: 1186 REGISTER

Supported: X-cisco-cme-sis-1.0.0

Supported: X-cisco-srtp-fallback

Contact: <sip:384@192.168.200.6:5060>;expires=600

Expires: 600

Content-Type: application/x-cisco-remotecc-response+xml

Content-Length: 188

<?xml version="1.0" encoding="UTF-8"?><x-cisco-remotecc-response><response><code>200</code><optionsind><cmeoption><b2bconf/></cmeoption></optionsind></response></x-cisco-remotecc-response>

IPPBX-CorpDesRancagua#

R0g22 · ‎09-05-2018

Enable the following for me -

debug ccsip message
debug ccsip error
debug voice ccapi inout

Make a test call, collect them in a text file and attach.

mcastillocandia · ‎09-14-2018

Hello

Sorry for a late response. I attach here what you asked for.

Thanks a lot for your help!

R0g22 · ‎09-14-2018

A single file please with all the debugs enabled.

mcastillocandia · ‎09-19-2018

Hi.

I attach here what you asked for.

Thanks a lot for your help!

R0g22 · ‎09-20-2018

Can you share ANI, DNIS and RDNIS info please ?

tfsnchez88 · ‎01-03-2019

I have the same problem, so could you solve this issue?

ashok_boin · ‎01-03-2019

Hi,

Have you tried contacting ITSP (your provider) about this error? Generally, ITSP expects "Early offer (SDP in in initial invite) but in your case SDP is also there in your 1st Invite.

Regards...

Ashok.

With best regards...
Ashok

Sreekanth Narayanan · ‎01-03-2019

Codec should not be a problem because the ITSP accepted the initial INVITE with g729.

The ITSP may be disconnecting the call because of the a=sendonly attribute in the INVITE sent to them when pressing Transfer.

Try adding this command on the router.

voice service voip

sip

midcall-signaling block

This will block the INVITE for sendonly being sent to the ITSP. The ITSP leg will always remain connected and not change states.

tfsnchez88 · ‎02-07-2019

I have the same problem, I issue this command but not work. So my problem its intermittent, before two or tree calls, the next one it drop.

Faisal Mohammed · ‎03-20-2019

Dear

Thank you so much this command help me more.

Regards

Faisal

Faisal Mohammed · ‎03-20-2019

Dear
Sreekanth Narayanan
Thank you so much this command help me more.

Regards
Faisal

MxHadi · ‎05-17-2024

thanks dear @Sreekanth Narayanan

we had this issue too

and your solution is working for us..

-- if this helped, please rate by click (Accept as solution) or (Helpful) --