Re: Want to forward the incoming call to Mobile number or international call - Page 2

mohammad saeed · ‎03-25-2021

Hi Guys,

I have CUCM 8.5 with SIP trunk from SP and I have CUBE as well.

I want if some one call internal extension and there is no answer then the call forward to mobile number or international Call, Can I configure this option inside CUCM? DO I need to do something in CUBE?

If there is any configuration document can help?

Thanks for all

Mohammad Saeed

TONY SMITH · ‎03-26-2021

OK and when you make an outbound call, how does your calling number get presented? I looked at a configuration using STC in Al Khobar, and it looks that we have that configured for full National numbers including the prefix, ie for Al Khobar the numbers are presented as 013xxxxxxx for calling party. Although inbound we're getting 7 digits. However I can't see that we've had to configure anything special for call forward in that case.

In the CUCM trunk we have Calling Party Selection set to "Originator". I think you may have yours set to one of the "redirect" options.

Can you show us a normal working Invite for an inbound and for an outbound call?

mohammad saeed · ‎03-26-2021

This is Outbound Call to 0112359535:

Mar 26 16:20:33.415: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0112359535@10.60.2.12:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3df842f689
From: <sip:6152@10.60.2.10>;tag=190489~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762520
To: <sip:0112359535@10.60.2.12>
Date: Fri, 26 Mar 2021 16:17:10 GMT
Call-ID: b6274c80-5e10906-1a3b4-a023c0a@10.60.2.10
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 3056028800-0000065536-0000007593-0167918602
Session-Expires: 1800
P-Asserted-Identity: <sip:6152@10.60.2.10>
Remote-Party-ID: <sip:6152@10.60.2.10>;party=calling;screen=yes;privacy=off
Contact: <sip:6152@10.60.2.10:5060>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 210

v=0
o=CiscoSystemsCCM-SIP 190489 1 IN IP4 10.60.2.10
s=SIP Call
c=IN IP4 10.60.2.11
t=0 0
m=audio 26186 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Mar 26 16:20:33.423: //4379312/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3df842f689
From: <sip:6152@10.60.2.10>;tag=190489~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762520
To: <sip:0112359535@10.60.2.12>
Date: Fri, 26 Mar 2021 16:20:33 GMT
Call-ID: b6274c80-5e10906-1a3b4-a023c0a@10.60.2.10
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Mar 26 16:20:33.427: //4379313/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0112359535@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5C0181E82
Remote-Party-ID: <sip:5386152@10.189.2.238>;party=calling;screen=yes;privacy=off
From: <sip:5386152@10.189.2.238>;tag=C0F6CF34-5D8
To: <sip:0112359535@10.200.7.157>
Date: Fri, 26 Mar 2021 16:20:33 GMT
Call-ID: 6356B08-8D8611EB-BCABDA71-29E28895@10.189.2.238
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3056028800-0000065536-0000007593-0167918602
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1616775633
Contact: <sip:5386152@10.189.2.238:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 296

v=0
o=CiscoSystemsSIP-GW-UserAgent 9615 3887 IN IP4 10.189.2.238
s=SIP Call
c=IN IP4 10.189.2.238
t=0 0
m=audio 18636 RTP/AVP 8 100 101
c=IN IP4 10.189.2.238
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=ptime:20

Mar 26 16:20:33.467: //4379313/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5C0181E82
Call-ID: 6356B08-8D8611EB-BCABDA71-29E28895@10.189.2.238
From: <sip:5386152@10.189.2.238>;tag=C0F6CF34-5D8
To: <sip:0112359535@10.200.7.157>
CSeq: 101 INVITE
Content-Length: 0

Mar 26 16:20:34.223: //4379313/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5C0181E82
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Call-ID: 6356B08-8D8611EB-BCABDA71-29E28895@10.189.2.238
From: <sip:5386152@10.189.2.238>;tag=C0F6CF34-5D8
To: <sip:0112359535@10.200.7.157>;tag=sbc0802pufbasak-CC-44
CSeq: 101 INVITE
Contact: <sip:0112359535@10.200.7.157:5060;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Require: 100rel
RSeq: 1
Content-Length: 193
Content-Type: application/sdp

v=0
o=- 15062868 15062868 IN IP4 10.200.7.157
s=SBC call
c=IN IP4 10.200.7.157
t=0 0
m=audio 37810 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Mar 26 16:20:34.227: //4379313/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
PRACK sip:0112359535@10.200.7.157:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5C01912B7
From: <sip:5386152@10.189.2.238>;tag=C0F6CF34-5D8
To: <sip:0112359535@10.200.7.157>;tag=sbc0802pufbasak-CC-44
Date: Fri, 26 Mar 2021 16:20:33 GMT
Call-ID: 6356B08-8D8611EB-BCABDA71-29E28895@10.189.2.238
CSeq: 102 PRACK
RAck: 1 101 INVITE
Route: <sip:10.200.7.157:5060;transport=udp;lr>
Allow-Events: telephone-event
Max-Forwards: 70
Content-Length: 0

Mar 26 16:20:34.227: //4379312/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3df842f689
From: <sip:6152@10.60.2.10>;tag=190489~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762520
To: <sip:0112359535@10.60.2.12>;tag=C0F6D258-2704
Date: Fri, 26 Mar 2021 16:20:33 GMT
Call-ID: b6274c80-5e10906-1a3b4-a023c0a@10.60.2.10
CSeq: 101 INVITE
Require: 100rel
RSeq: 7776
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:0112359535@10.60.2.12>;party=called;screen=no;privacy=off
Contact: <sip:0112359535@10.60.2.12:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 241

v=0
o=CiscoSystemsSIP-GW-UserAgent 8108 9300 IN IP4 10.60.2.12
s=SIP Call
c=IN IP4 10.60.2.12
t=0 0
m=audio 32272 RTP/AVP 8 101
c=IN IP4 10.60.2.12
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Mar 26 16:20:34.231: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:0112359535@10.60.2.12:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3e02e528f06
From: <sip:6152@10.60.2.10>;tag=190489~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762520
To: <sip:0112359535@10.60.2.12>;tag=C0F6D258-2704
Date: Fri, 26 Mar 2021 16:17:10 GMT
Call-ID: b6274c80-5e10906-1a3b4-a023c0a@10.60.2.10
CSeq: 102 PRACK
RAck: 7776 101 INVITE
Allow-Events: presence
Max-Forwards: 70
Content-Length: 0

Mar 26 16:20:34.231: //4379312/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3e02e528f06
From: <sip:6152@10.60.2.10>;tag=190489~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762520
To: <sip:0112359535@10.60.2.12>;tag=C0F6D258-2704
Date: Fri, 26 Mar 2021 16:20:34 GMT
Call-ID: b6274c80-5e10906-1a3b4-a023c0a@10.60.2.10
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 PRACK
Content-Length: 0

Mar 26 16:20:34.283: //4379313/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5C01912B7
Call-ID: 6356B08-8D8611EB-BCABDA71-29E28895@10.189.2.238
From: <sip:5386152@10.189.2.238>;tag=C0F6CF34-5D8
To: <sip:0112359535@10.200.7.157>;tag=sbc0802pufbasak-CC-44
CSeq: 102 PRACK
Content-Length: 0

Mar 26 16:20:37.035: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.189.2.238:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKfbdpkdut7727saa7skbohodcaT41528
Call-ID: isbckeppb2kfo7ddkk2pc2pokptc7cc7deo7@SoftX3000
From: <sip:10.189.2.238:5060>;tag=sbc0806cdtdspfk
To: <sip:10.189.2.238>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

Mar 26 16:20:37.035: //4379314/085C8FC2BCAC/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKfbdpkdut7727saa7skbohodcaT41528
From: <sip:10.189.2.238:5060>;tag=sbc0806cdtdspfk
To: <sip:10.189.2.238>;tag=C0F6DD50-2186
Date: Fri, 26 Mar 2021 16:20:37 GMT
Call-ID: isbckeppb2kfo7ddkk2pc2pokptc7cc7deo7@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 166

v=0
o=CiscoSystemsSIP-GW-UserAgent 4306 170 IN IP4 10.189.2.238
s=SIP Call
c=IN IP4 10.189.2.238
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 10.189.2.238

Mar 26 16:20:40.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:0112359535@10.60.2.12:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3df842f689
From: <sip:6152@10.60.2.10>;tag=190489~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762520
To: <sip:0112359535@10.60.2.12>
Date: Fri, 26 Mar 2021 16:17:10 GMT
Call-ID: b6274c80-5e10906-1a3b4-a023c0a@10.60.2.10
CSeq: 101 CANCEL
Max-Forwards: 70
Content-Length: 0

Mar 26 16:20:40.354: //4379312/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3df842f689
From: <sip:6152@10.60.2.10>;tag=190489~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762520
To: <sip:0112359535@10.60.2.12>
Date: Fri, 26 Mar 2021 16:20:40 GMT
Call-ID: b6274c80-5e10906-1a3b4-a023c0a@10.60.2.10
CSeq: 101 CANCEL
Content-Length: 0

Mar 26 16:20:40.354: //4379313/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:0112359535@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5C0181E82
From: <sip:5386152@10.189.2.238>;tag=C0F6CF34-5D8
To: <sip:0112359535@10.200.7.157>
Date: Fri, 26 Mar 2021 16:20:33 GMT
Call-ID: 6356B08-8D8611EB-BCABDA71-29E28895@10.189.2.238
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1616775640
Reason: Q.850;cause=16
Content-Length: 0

Mar 26 16:20:40.354: //4379312/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3df842f689
From: <sip:6152@10.60.2.10>;tag=190489~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762520
To: <sip:0112359535@10.60.2.12>;tag=C0F6D258-2704
Date: Fri, 26 Mar 2021 16:20:40 GMT
Call-ID: b6274c80-5e10906-1a3b4-a023c0a@10.60.2.10
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0

Mar 26 16:20:40.354: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0112359535@10.60.2.12:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3df842f689
From: <sip:6152@10.60.2.10>;tag=190489~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762520
To: <sip:0112359535@10.60.2.12>;tag=C0F6D258-2704
Date: Fri, 26 Mar 2021 16:17:10 GMT
Call-ID: b6274c80-5e10906-1a3b4-a023c0a@10.60.2.10
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

Mar 26 16:20:40.354: //4379312/B6274C800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2BCB1AF8
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 6152
Called Number : 0112359535
Source IP Address (Sig 10.60.2.12
Destn SIP Req Addr:Port : 10.60.2.10:5060
Destn SIP Resp Addr:Port : 10.60.2.10:5060
Destination Name : 10.60.2.10

Mar 26 16:20:40.354: //4379312/B6274C800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.60.2.12
Source IP Port (Media): 32272
Destn IP Address (Media): 10.60.2.11
Destn IP Port (Media): 26186
Orig Destn IP Address:Port (Media): [ - ]:0

Mar 26 16:20:40.354: //4379312/B6274C800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 487

Mar 26 16:20:40.402: //4379313/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5C0181E82
Call-ID: 6356B08-8D8611EB-BCABDA71-29E28895@10.189.2.238
From: <sip:5386152@10.189.2.238>;tag=C0F6CF34-5D8
To: <sip:0112359535@10.200.7.157>
CSeq: 101 CANCEL
Content-Length: 0

Mar 26 16:20:40.430: //4379313/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5C0181E82
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Call-ID: 6356B08-8D8611EB-BCABDA71-29E28895@10.189.2.238
From: <sip:5386152@10.189.2.238>;tag=C0F6CF34-5D8
To: <sip:0112359535@10.200.7.157>;tag=sbc0802pufbasak-CC-44
CSeq: 101 INVITE
Warning: 399 SoftX3000 "SS030001F00156L00817[00000] Cancel received from network"
Content-Length: 0

Mar 26 16:20:40.430: //4379313/B6274C800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2BCEF848
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 5386152
Called Number : 0112359535
Source IP Address (Sig 10.189.2.238
Destn SIP Req Addr:Port : 10.200.7.157:5060
Destn SIP Resp Addr:Port : 10.200.7.157:5060
Destination Name : 10.200.7.157

Mar 26 16:20:40.430: //4379313/B6274C800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.189.2.238
Source IP Port (Media): 18636
Destn IP Address (Media): 10.200.7.157
Destn IP Port (Media): 37810
Orig Destn IP Address:Port (Media): [ - ]:0

Mar 26 16:20:40.430: //4379313/B6274C800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 487

Mar 26 16:20:40.430: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:0112359535@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5C0181E82
From: <sip:5386152@10.189.2.238>;tag=C0F6CF34-5D8
To: <sip:0112359535@10.200.7.157>;tag=sbc0802pufbasak-CC-44
Date: Fri, 26 Mar 2021 16:20:33 GMT
Call-ID: 6356B08-8D8611EB-BCABDA71-29E28895@10.189.2.238
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

And this is the inbound call from 0566601998 to my extension 6152 without any forwarding :

Mar 26 16:22:57.464: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:5386152@10.189.2.238;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKutdpossdese4ptdsobu2kadtoT05336
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Call-ID: isbcstuftatfdtdbuktsfs2ut7ehoh47kf2d@SoftX3000
From: <sip:566601998@10.189.2.238;user=phone>;tag=sbc0806s2tobset-CC-34
To: <sip:5386152@10.189.2.238;user=phone>
CSeq: 1 INVITE
Max-Forwards: 69
Contact: <sip:566601998@10.200.7.157:5060;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R010
Supported: 100rel
Content-Length: 319
Content-Type: application/sdp

v=0
o=- 15710774 15710774 IN IP4 10.200.7.157
s=SBC call
c=IN IP4 10.200.7.157
t=0 0
m=audio 44366 RTP/AVP 8 102 102 0 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:102 AMR/8000
a=rtpmap:102 AMR/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-15
a=fmtp:18 annexb=yes

Mar 26 16:22:57.468: //4379330/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKutdpossdese4ptdsobu2kadtoT05336
From: <sip:566601998@10.189.2.238;user=phone>;tag=sbc0806s2tobset-CC-34
To: <sip:5386152@10.189.2.238;user=phone>
Date: Fri, 26 Mar 2021 16:22:57 GMT
Call-ID: isbcstuftatfdtdbuktsfs2ut7ehoh47kf2d@SoftX3000
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Mar 26 16:22:57.468: //4379331/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:6152@10.60.2.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.2.12:5060;branch=z9hG4bK5C01D712
Remote-Party-ID: <sip:566601998@10.60.2.12>;party=calling;screen=no;privacy=off
From: <sip:566601998@10.60.2.12>;tag=C0F901E4-9FD
To: <sip:6152@10.60.2.10>
Date: Fri, 26 Mar 2021 16:22:57 GMT
Call-ID: 5C1113D9-8D8611EB-BCCCDA71-29E28895@10.60.2.12
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1544583089-2374373867-3167148657-0702711957
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1616775777
Contact: <sip:566601998@10.60.2.12:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 290

v=0
o=CiscoSystemsSIP-GW-UserAgent 5156 5008 IN IP4 10.60.2.12
s=SIP Call
c=IN IP4 10.60.2.12
t=0 0
m=audio 30678 RTP/AVP 8 100 101
c=IN IP4 10.60.2.12
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=ptime:20

Mar 26 16:22:57.472: //4379331/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.60.2.12:5060;branch=z9hG4bK5C01D712
From: <sip:566601998@10.60.2.12>;tag=C0F901E4-9FD
To: <sip:6152@10.60.2.10>
Date: Fri, 26 Mar 2021 16:19:34 GMT
Call-ID: 5C1113D9-8D8611EB-BCCCDA71-29E28895@10.60.2.12
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0

Mar 26 16:22:57.476: //4379331/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.60.2.12:5060;branch=z9hG4bK5C01D712
From: <sip:566601998@10.60.2.12>;tag=C0F901E4-9FD
To: <sip:6152@10.60.2.10>;tag=190497~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762568
Date: Fri, 26 Mar 2021 16:19:34 GMT
Call-ID: 5C1113D9-8D8611EB-BCCCDA71-29E28895@10.60.2.12
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
P-Asserted-Identity: <sip:6152@10.60.2.10>
Remote-Party-ID: <sip:6152@10.60.2.10>;party=called;screen=yes;privacy=off
Contact: <sip:6152@10.60.2.10:5060>
Content-Length: 0

Mar 26 16:22:57.476: //4379330/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKutdpossdese4ptdsobu2kadtoT05336
From: <sip:566601998@10.189.2.238;user=phone>;tag=sbc0806s2tobset-CC-34
To: <sip:5386152@10.189.2.238;user=phone>;tag=C0F901EC-1793
Date: Fri, 26 Mar 2021 16:22:57 GMT
Call-ID: isbcstuftatfdtdbuktsfs2ut7ehoh47kf2d@SoftX3000
CSeq: 1 INVITE
Require: 100rel
RSeq: 4759
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:6152@10.189.2.238>;party=called;screen=yes;privacy=off
Contact: <sip:6152@10.189.2.238:5060>
Record-Route: <sip:10.200.7.157:5060;transport=udp;lr>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Mar 26 16:22:57.508: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.189.2.238:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKdkfdbukhtp4pooadkbefbofeeT01024
Call-ID: isbcedshkbepebs7dspfoepft2up477debuu@SoftX3000
From: <sip:10.189.2.238:5060>;tag=sbc08042b2ofcfc
To: <sip:10.189.2.238>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

Mar 26 16:22:57.512: //4379332/5C17CA29BCCD/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKdkfdbukhtp4pooadkbefbofeeT01024
From: <sip:10.189.2.238:5060>;tag=sbc08042b2ofcfc
To: <sip:10.189.2.238>;tag=C0F90210-1989
Date: Fri, 26 Mar 2021 16:22:57 GMT
Call-ID: isbcedshkbepebs7dspfoepft2up477debuu@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 166

v=0
o=CiscoSystemsSIP-GW-UserAgent 6600 268 IN IP4 10.189.2.238
s=SIP Call
c=IN IP4 10.189.2.238
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 10.189.2.238

Mar 26 16:22:57.612: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
PRACK sip:6152@10.189.2.238:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKohdthbspussd2uabbscph2b2aT05340
Call-ID: isbcstuftatfdtdbuktsfs2ut7ehoh47kf2d@SoftX3000
From: <sip:566601998@10.189.2.238;user=phone>;tag=sbc0806s2tobset-CC-34
To: <sip:5386152@10.189.2.238;user=phone>;tag=C0F901EC-1793
CSeq: 2 PRACK
Max-Forwards: 70
RAck: 4759 1 INVITE
Content-Length: 0

Mar 26 16:22:57.612: //4379330/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKohdthbspussd2uabbscph2b2aT05340
From: <sip:566601998@10.189.2.238;user=phone>;tag=sbc0806s2tobset-CC-34
To: <sip:5386152@10.189.2.238;user=phone>;tag=C0F901EC-1793
Date: Fri, 26 Mar 2021 16:22:57 GMT
Call-ID: isbcstuftatfdtdbuktsfs2ut7ehoh47kf2d@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 2 PRACK
Content-Length: 0

Mar 26 16:23:01.048: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.60.2.12:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3e42c37432d
From: <sip:10.60.2.10>;tag=1856495330
To: <sip:10.60.2.12>
Date: Fri, 26 Mar 2021 16:19:38 GMT
Call-ID: e5e4e80-5e1099a-1a3b8-a023c0a@10.60.2.10
User-Agent: Cisco-CUCM8.6
CSeq: 101 OPTIONS
Contact: <sip:10.60.2.10:5060>
Max-Forwards: 0
Content-Length: 0

Mar 26 16:23:01.048: //4379333/5E335622BCCE/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.2.10:5060;branch=z9hG4bK1e3e42c37432d
From: <sip:10.60.2.10>;tag=1856495330
To: <sip:10.60.2.12>;tag=C0F90FE0-F6A
Date: Fri, 26 Mar 2021 16:23:01 GMT
Call-ID: e5e4e80-5e1099a-1a3b8-a023c0a@10.60.2.10
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 161

v=0
o=CiscoSystemsSIP-GW-UserAgent 6400 1046 IN IP4 10.60.2.12
s=SIP Call
c=IN IP4 10.60.2.12
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 10.60.2.12

Mar 26 16:23:02.012: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:5386152@10.189.2.238;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKutdpossdese4ptdsobu2kadtoT05336
Call-ID: isbcstuftatfdtdbuktsfs2ut7ehoh47kf2d@SoftX3000
From: <sip:566601998@10.189.2.238;user=phone>;tag=sbc0806s2tobset-CC-34
To: <sip:5386152@10.189.2.238;user=phone>
CSeq: 1 CANCEL
Reason: Q.850;cause=16;text="normal call clearing"
Max-Forwards: 70
Content-Length: 0

Mar 26 16:23:02.012: //4379330/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKutdpossdese4ptdsobu2kadtoT05336
From: <sip:566601998@10.189.2.238;user=phone>;tag=sbc0806s2tobset-CC-34
To: <sip:5386152@10.189.2.238;user=phone>
Date: Fri, 26 Mar 2021 16:23:02 GMT
Call-ID: isbcstuftatfdtdbuktsfs2ut7ehoh47kf2d@SoftX3000
CSeq: 1 CANCEL
Content-Length: 0

Mar 26 16:23:02.012: //4379331/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:6152@10.60.2.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.2.12:5060;branch=z9hG4bK5C01D712
From: <sip:566601998@10.60.2.12>;tag=C0F901E4-9FD
To: <sip:6152@10.60.2.10>
Date: Fri, 26 Mar 2021 16:22:57 GMT
Call-ID: 5C1113D9-8D8611EB-BCCCDA71-29E28895@10.60.2.12
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1616775782
Reason: Q.850;cause=16
Content-Length: 0

Mar 26 16:23:02.012: //4379330/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKutdpossdese4ptdsobu2kadtoT05336
From: <sip:566601998@10.189.2.238;user=phone>;tag=sbc0806s2tobset-CC-34
To: <sip:5386152@10.189.2.238;user=phone>;tag=C0F901EC-1793
Date: Fri, 26 Mar 2021 16:23:02 GMT
Call-ID: isbcstuftatfdtdbuktsfs2ut7ehoh47kf2d@SoftX3000
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0

Mar 26 16:23:02.016: //4379331/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.2.12:5060;branch=z9hG4bK5C01D712
From: <sip:566601998@10.60.2.12>;tag=C0F901E4-9FD
To: <sip:6152@10.60.2.10>
Date: Fri, 26 Mar 2021 16:19:39 GMT
Call-ID: 5C1113D9-8D8611EB-BCCCDA71-29E28895@10.60.2.12
CSeq: 101 CANCEL
Content-Length: 0

Mar 26 16:23:02.016: //4379331/5C1077B1BCC6/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.60.2.12:5060;branch=z9hG4bK5C01D712
From: <sip:566601998@10.60.2.12>;tag=C0F901E4-9FD
To: <sip:6152@10.60.2.10>;tag=190497~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762568
Date: Fri, 26 Mar 2021 16:19:39 GMT
Call-ID: 5C1113D9-8D8611EB-BCCCDA71-29E28895@10.60.2.12
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0

Mar 26 16:23:02.016: //4379331/5C1077B1BCC6/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2BCAC108
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 566601998
Called Number : 6152
Source IP Address (Sig 10.60.2.12
Destn SIP Req Addr:Port : 10.60.2.10:5060
Destn SIP Resp Addr:Port : 10.60.2.10:5060
Destination Name : 10.60.2.10

Mar 26 16:23:02.016: //4379331/5C1077B1BCC6/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 10.60.2.12
Source IP Port (Media): 30678
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

Mar 26 16:23:02.016: //4379331/5C1077B1BCC6/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 487

Mar 26 16:23:02.016: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:6152@10.60.2.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.60.2.12:5060;branch=z9hG4bK5C01D712
From: <sip:566601998@10.60.2.12>;tag=C0F901E4-9FD
To: <sip:6152@10.60.2.10>;tag=190497~89711f2a-613c-4c32-92da-cd0f7d30bcbc-20762568
Date: Fri, 26 Mar 2021 16:22:57 GMT
Call-ID: 5C1113D9-8D8611EB-BCCCDA71-29E28895@10.60.2.12
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

Mar 26 16:23:02.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:5386152@10.189.2.238;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKutdpossdese4ptdsobu2kadtoT05336
Call-ID: isbcstuftatfdtdbuktsfs2ut7ehoh47kf2d@SoftX3000
From: <sip:566601998@10.189.2.238;user=phone>;tag=sbc0806s2tobset-CC-34
To: <sip:5386152@10.189.2.238;user=phone>;tag=C0F901EC-1793
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

Mar 26 16:23:02.044: //4379330/5C1077B1BCC6/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2BCB1AF8
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 566601998
Called Number : 5386152
Source IP Address (Sig 10.189.2.238
Destn SIP Req Addr:Port : 10.200.7.157:5060
Destn SIP Resp Addr:Port : 10.200.7.157:5060
Destination Name : 10.200.7.157

Mar 26 16:23:02.044: //4379330/5C1077B1BCC6/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 116 (tx), 116 (rx)
Source IP Address (Media): 10.189.2.238
Source IP Port (Media): 18710
Destn IP Address (Media): 10.200.7.157
Destn IP Port (Media): 44366
Orig Destn IP Address:Port (Media): [ - ]:0

Mar 26 16:23:02.044: //4379330/5C1077B1BCC6/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 487

Roger Kallberg · ‎03-28-2021

I might be blind, but as far as I can tell you are not sending any diversion header.

TONY SMITH · ‎03-29-2021

My head's spinning a little as well. The last debug (Mar 26 16:22:57) was for a normal outgoing call which I wanted to see to compare headers and number formats. The redirected call shared earlier shows a Diversion header with the number in four digit form ...

Mar 26 14:44:57.909: //4378760/5BD3BF000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0112359535@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5BFCB3A0
Remote-Party-ID: <sip:5386152@10.189.2.238>;party=calling;screen=yes;privacy=off
From: <sip:5386152@10.189.2.238>;tag=C09F4A84-1829
To: <sip:0112359535@10.200.7.157>
Date: Fri, 26 Mar 2021 14:44:57 GMT
Call-ID: AB9372E6-8D7811EB-B9FBDA71-29E28895@10.189.2.238
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1540603648-0000065536-0000007571-0167918602
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1616769897
Contact: <sip:5386152@10.189.2.238:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 66
Diversion: <sip:6152@10.60.2.10>;privacy=off;reason=unconditional;screen=yes
Session-Expires: 1800
Content-Type: application/sdp

If this was AT&T I would not be happy with that header as they required a Diversion header to contain a valid DDI.

However I don't have any documentation or notes on STC's special requirements for redirected calls. So in the first instance I think it would be helpful to get the call working, even though it might not be displaying the OP's preferred calling number.

Roger Kallberg · ‎03-29-2021

Your right Tony, there was a diversion header in the earlier shared debug output, I didn't even look past the most recent one.

Mohammad if you want to modify the content of the diversion header in the router you could do it with this.

 request INVITE sip-header Diversion modify "Diversion:(.*)(<sip:)(.*)(@.*>)" "Diversion: \2<number in format required by SP, could be +E.164>\4" 
 request INVITE sip-header Diversion add "Diversion: <sip:<number in format required by SP, could be +E.164>@<IP for outside interface on SBC>>"

Modify to fit your needs and add it to you SIP profile attached to the outbound dial peer towards your service provider.

mohammad saeed · ‎03-29-2021

Thanks Roger, Thanks Tony, I will try to change the header because it makes confused the header are the same with both but with forwarder is not making the call!

mohammad saeed · ‎03-29-2021

if suppose the caller is "A" and the IP phone number is "B" and the forwarder number is "C", which number I have to put it in SIP profile command:

 request INVITE sip-header Diversion modify "Diversion:(.*)(<sip:)(.*)(@.*>)" "Diversion: \2<"B">\4" 
 request INVITE sip-header Diversion add "Diversion: <sip:<"C">@<IP for outside interface on SBC>>"

TONY SMITH · ‎03-29-2021

From my example it would number "B", the number the original caller dialled, but which is redirected outbound. The reason would be if the ITSP needs to match it to your account for billing or security purposes.

Here's an example of my workings from something similar. In that case we hard-coded the profile to the customer's main billing number as it wasn't worth the effort to convert all their four digit extensions into National format. That was North America so the XXX is a ten digit number. In your case it would be your seven digit, or maybe full national .. or maybe with leading zero .. or maybe E164 ..

CUCM
	Enable Diversion header on Trunk
CUBE
	voice class sip-profiles 9
	 request INVITE sip-header Diversion modify "<sip:(.*)@(.*)>" "<sip:XXXXXXXXXX@\2>"

	dial-peer voice 9 voip
	 voice-class sip profiles 9

However I really think we should try and get the call working without the Diversion header in the first instance. We don't know for sure that it the Diversion header causing the error, or the exact format that STC needs, or even if they support this method.

Roger Kallberg · ‎03-29-2021

As Tony wrote it's the DDI number that is called to, aka B number, that you should have in the diversion header. As also pointed out it is quite often enough to put the main number for the circuit as the number in the diversion header, so you should be able to hard code it to one specific number.

For the most I tend to pass numbers in E.164 format for any number fields passed in the signaling of the call, as this is from my experience for the most supported on SIP services.

mohammad saeed · ‎03-26-2021

I got these instruction from someone he is in STC:

While forwarding a call landing on Your specific sip number an invite msg should be initiated from your voice GW and calling number should be your specific SIP number and called party should be the number where you want to forward a call
Calling number should be 7digit and the B-party number should be any number within KSA but called party number format should be correct.

But don't know how to apply it in the gateway!

TONY SMITH · ‎03-28-2021

In the examples that you showed your are already presenting your number as the calling number, in seven digit format. I had a look at our Saudi customer and we do the same there, internally their numbers are presented in full format with regional code and leading zero (013xxxxxxx) but a rule in the gateway truncates that down to seven digits.

Your From header appears the same in your working outbound call ...

Mar 26 16:20:33.427: //4379313/B6274C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0112359535@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5C0181E82
Remote-Party-ID: <sip:5386152@10.189.2.238>;party=calling;screen=yes;privacy=off
From: <sip:5386152@10.189.2.238>;tag=C0F6CF34-5D8
To: <sip:0112359535@10.200.7.157>

And in your non-working redirected call ...

Mar 26 14:44:57.909: //4378760/5BD3BF000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0112359535@10.200.7.157:5060 SIP/2.0
Via: SIP/2.0/UDP 10.189.2.238:5060;branch=z9hG4bK5BFCB3A0
Remote-Party-ID: <sip:5386152@10.189.2.238>;party=calling;screen=yes;privacy=off
From: <sip:5386152@10.189.2.238>;tag=C09F4A84-1829
To: <sip:0112359535@10.200.7.157>

So it may still be down to the Diversion header and it's incomplete number. To start with let's remove that from your CUCM SIP trunk outbound settings, and also change "Calling party selection" to last redirected number if it's not already set.

To give a little background, in a normal call forward or redirection, you normally want the final called party to see the original caller. So let's say caller "A" calls your DDI "B" which you have forwarded to destination "C". Normally in this case you'd want the called party to see the original calling number, ie a call from "A" rather than a call from your DDI "B".

Fixing this up is where ITSPs differ wildly.

So if I guess correctly, with this initial change suggested today you should get your call forward working, but the final called party will see the calling number as your DDI rather than the original caller. If that works then we can take it from there. If not then post up a SIP debug as before.

Ibrahim AlQumairi · ‎04-13-2023

SNR Issues when calling from an EXTERNAL PSTN Number:
1. Please go to the H323 configuration page on CUCM and look for – “Call Routing Information – Outbound Calls” – let me know what you have under “Calling Party Selection” This parameter needs to be set to “Last Redirect Number”

2. Go to System – Service Parameter – select the publisher — Cisco CallManager (Active) and look for “Honor Gateway or Trunk Outbound Calling Party Selection for Mobile Connect Calls” – This parameter needs to be set to TRUE.

I have tried these two steps and the SNR works fine for internal and external calls

Also don't forget to disable the "Diversion Header" for Outbound Calls