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stillnick
Beginner

Problem in SIP-UA called Number given by SIP provider.

Hi. Does anyone know why my SIP Provider is not sending me the right called number? Instead it is sending me my Client ID.

 

This is the debug:

Feb 1 2021 16:08:37.740 AWST: //77372/8597BD2B82F2/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x3ECC1278, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=047XXXXX613,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=7656XX7(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=110, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)

 

My "called number" should be the number which the client called to and not my account ID.

 

 

This is my sip-ua conf:

sip-ua
 credentials username 76XXX7 password 7 124C5144XXXXXXX57B7272 realm sip.provider.com.au 
 authentication username 76XXX7 password 7 124C5144XXXXXXX57B7272 realm sip.provider.com.au
 retry invite 5
 retry bye 2
 retry cancel 2 
 retry register 3
 retry options 10
 timers trying 550
 timers expires 60000
 timers connect 100
 no timers hold
 timers register 1000
 registrar 1 dns:sip-west.mycloudpbx.com.au expires 180
 sip-server dns:sip-west.mycloudpbx.com.au
 registration spike 50
 reason-header override
 connection-reuse
 host-registrar

 

 

Any idea would be much appreciated.

 

Thanks

10 REPLIES 10

Hi, 

please see if you are using any normalization on your inbound dial peer? .If there are no SIP profiles in use, and if you are certain that your invite message shows the username instead of the called number, then please contact your ITSP provider with the details. 

 

The community may help to re-check your configuration in case you post it here. 

 

Regards, 

*** Rate Usefull posts 

Hi Shalidm thanks for replying to me.

 

I did all mentioned checks and I'm still getting my account ID instead of the phone number. SP provider tested SIP trunk in one of their router and it is working as expected. Here is another debug that shows this.

 

Feb 1 2021 19:07:45.732 AWST: //83879/8BD7ABC19915/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x2280FE90
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 047XXXX613
Called Number : 76XXX7
Source IP Address (Sig 10.1.100.200
Destn SIP Req Addr:Port : 10.1.200.200:5060
Destn SIP Resp Addr:Port : 10.1.200.200:5060
Destination Name : 10.1.200.200

 

 

Thanks

If you believe it’s your configuration that is at fault please post it here as an attached file so that we can verify it. It would also be of help if you can capture the output of debug ccsip message and debug voip ccapi inout in a separate file.

Please rate all useful posts

Hi all. Thanks again for the help.

 

Here will go 2 files with configuration and debug +  the debug message bellow.

 


Feb 2 2021 07:48:02.132 AWST: //96820/C1621E778782/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK103751E16
From: <sip:0477770613@10.1.100.200>;tag=15EF1274-1325
To: <sip:76XX17@10.1.200.200>
Date: Mon, 01 Feb 2021 23:48:02 GMT
Call-ID: C163XXC7-641E11EB-8788FE72-5C880660@10.1.100.200
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0


Feb 2 2021 07:48:02.188 AWST: //96820/C1621E778782/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK103751E16
From: <sip:0477770613@10.1.100.200>;tag=15EF1274-1325
To: <sip:76XX17@10.1.200.200>;tag=25111894~7bae6435-a194-400a-8806-8ddaeecb994f-19053275
Date: Mon, 01 Feb 2021 23:48:02 GMT
Call-ID: C163XXC7-641E11EB-8788FE72-5C880660@10.1.100.200
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Server: Cisco-CUCM11.5
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP;x-cisco-qos-tcl=true
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-ID: 41822e7000105000a000007686fd9d10;remote=63ea10d8e01b969ce22c87ab25111894
P-Asserted-Identity: "Interite Main Line" <sip:633@10.1.200.200>
Remote-Party-ID: "Interite Main Line" <sip:633@10.1.200.200>;party=called;screen=yes;privacy=off
Contact: <sip:76XX17@10.1.200.200:5060>;+u.sip!devicename.ccm.cisco.com="SEP007686FD9D10";video
Content-Length: 0


Feb 2 2021 07:48:02.188 AWST: //96819/C1621E778782/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKr38Dpac2tQrFB
From: <sip:0477770613@27.127.233.78>;tag=1vXjD0ZgXjQrm
To: <sip:0893540105@sip>;tag=15EF12B0-8F6
Date: Mon, 01 Feb 2021 23:48:02 GMT
Call-ID: c32826da-df8a-1239-eabe-00163ea67143
CSeq: 31557849 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:76XX17@10.1.222.2:5060>
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0

terminal mon
Feb 2 2021 07:48:03.720 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:76XX17@10.1.222.2:5060 SIP/2.0
v:SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKr38Dpac2tQrFB
Route:<sip:76XX17@203.54.247.182:20704>
Max-Forwards:65
f:<sip:0477770613@27.127.233.78>;tag=1vXjD0ZgXjQrm
t:<sip:0893540105@sip>
i:c32826da-df8a-1239-eabe-00163ea67143
CSeq:31557849 CANCEL
Reason:Q.850;cause=16;text=""
l:0


Feb 2 2021 07:48:03.720 AWST: //96819/C1621E778782/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKr38Dpac2tQrFB
From: <sip:0477770613@27.127.233.78>;tag=1vXjD0ZgXjQrm
To: <sip:0893540105@sip>
Date: Mon, 01 Feb 2021 23:48:03 GMT
Call-ID: c32826da-df8a-1239-eabe-00163ea67143
CSeq: 31557849 CANCEL
Content-Length: 0


Feb 2 2021 07:48:03.724 AWST: //96820/C1621E778782/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:76XX17@10.1.200.200:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK103751E16
From: <sip:0477770613@10.1.100.200>;tag=15EF1274-1325
To: <sip:76XX17@10.1.200.200>
Date: Mon, 01 Feb 2021 23:48:02 GMT
Call-ID: C163XXC7-641E11EB-8788FE72-5C880660@10.1.100.200
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1612223283
Reason: Q.850;cause=16
Content-Length: 0


Feb 2 2021 07:48:03.724 AWST: //96819/C1621E778782/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKr38Dpac2tQrFB
From: <sip:0477770613@27.127.233.78>;tag=1vXjD0ZgXjQrm
To: <sip:0893540105@sip>;tag=15EF12B0-8F6
Date: Mon, 01 Feb 2021 23:48:03 GMT
Call-ID: c32826da-df8a-1239-eabe-00163ea67143
CSeq: 31557849 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=16
Content-Length: 0


Feb 2 2021 07:48:03.724 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:76XX17@10.1.222.2:5060 SIP/2.0
v:SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKr38Dpac2tQrFB
Route:<sip:76XX17@203.54.247.182:20704>
Max-Forwards:65
f:<sip:0477770613@27.127.233.78>;tag=1vXjD0ZgXjQrm
t:<sip:0893540105@sip>;tag=15EF12B0-8F6
i:c32826da-df8a-1239-eabe-00163ea67143
CSeq:31557849 ACK
l:0


Feb 2 2021 07:48:03.728 AWST: //96820/C1621E778782/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK103751E16
From: <sip:0477770613@10.1.100.200>;tag=15EF1274-1325
To: <sip:76XX17@10.1.200.200>
Date: Mon, 01 Feb 2021 23:48:03 GMT
Call-ID: C163XXC7-641E11EB-8788FE72-5C880660@10.1.100.200
Server: Cisco-CUCM11.5
CSeq: 101 CANCEL
Content-Length: 0


Feb 2 2021 07:48:03.728 AWST: //96820/C1621E778782/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK103751E16
From: <sip:0477770613@10.1.100.200>;tag=15EF1274-1325
To: <sip:76XX17@10.1.200.200>;tag=25111894~7bae6435-a194-400a-8806-8ddaeecb994f-19053275
Date: Mon, 01 Feb 2021 23:48:03 GMT
Call-ID: C163XXC7-641E11EB-8788FE72-5C880660@10.1.100.200
CSeq: 101 INVITE
Allow-Events: presence
Server: Cisco-CUCM11.5
Content-Length: 0


Feb 2 2021 07:48:03.728 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: no mon
ICI-AU-CNN-VRO-01#ACK sip:76XX17@10.1.200.200:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK103751E16
From: <sip:0477770613@10.1.100.200>;tag=15EF1274-1325
To: <sip:76XX17@10.1.200.200>;tag=25111894~7bae6435-a194-400a-8806-8ddaeecb994f-19053275
Date: Mon, 01 Feb 2021 23:48:02 GMT
Call-ID: C163XXC7-641E11EB-8788FE72-5C880660@10.1.100.200
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

 

The information is incomplete. 

The logs don't contain the first invite message and acknowledgment. 

The configuration also only included the SIP-UA. 

If the ITSP is sending the correct values to your gateway, then as per my understanding, the header manipulation (SIP normalization) might occurs either at your dial peers or any other device hope before it reaches your gateway.  We may have better clarity if we could get at least the complete set of logs for a single call. 

 

Regards, 

 

Hi there 

As mentioned in the previous post and by @Roger Kallberg, could you please attach your configuration and the output of debug ccsip message.  So we may able to help you further.  

Regards, 

 

Sorry Shalid, I must have missed something. Here will go the full debug.

 

Full conf file attached as well.

 


Feb 2 2021 08:46:00.363 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
NOTIFY sip:765617@10.1.222.2:5060 SIP/2.0
v:SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKrHU4aKaBXNUSe
Route:<sip:765617@203.54.247.182:20704>
Max-Forwards:70
f:sip:765617@sip;tag=H70NvUcD9mU7N
t:sip:765617@sip
i:dc5ab332-df92-1239-eabe-00163ea67143
CSeq:31559588 NOTIFY
m:sip:mod_sofia@27.127.233.78:5060
User-Agent:HV
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
k:timer,path,replaces
o:message-summary
u:talk,hold,conference,refer
Subscription-State:terminated;reason=noresource
c:application/simple-message-summary
l:57

Messages-Waiting: no
Message-Account: sip:765617@sip


Feb 2 2021 08:46:00.511 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:100@203.54.247.182 SIP/2.0
Via: SIP/2.0/UDP 193.29.14.109:5080;branch=z9hG4bK-2417934288;rport
Content-Length: 0
From: "sipvicious"<sip:100@1.1.1.1>;tag=6362333666376236313363340132343632303534323632
Accept: application/sdp
User-Agent: friendly-scanner
To: "sipvicious"<sip:100@1.1.1.1>
Contact: sip:100@193.29.14.109:5080
CSeq: 1 OPTIONS
Call-ID: 537383487426712960209696
Max-Forwards: 70


Feb 2 2021 08:46:00.511 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid IP Address'
Via: SIP/2.0/UDP 193.29.14.109:5080;branch=z9hG4bK-2417934288;rport
From: "sipvicious"<sip:100@1.1.1.1>;tag=6362333666376236313363340132343632303534323632
To: "sipvicious"<sip:100@1.1.1.1>;tag=162425DC-4F3
Date: Tue, 02 Feb 2021 00:46:00 GMT
Call-ID: 537383487426712960209696
Server: Cisco-SIPGateway/IOS-15.4.3.M3
CSeq: 1 OPTIONS
Content-Length: 0


Feb 2 2021 08:46:01.955 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:765617@10.1.222.2:5060 SIP/2.0
v:SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKvN07HZDSHSm4c
Route:<sip:765617@203.54.247.182:20704>
Max-Forwards:65
f:<sip:0477770613@27.127.233.78>;tag=SFBvamK9HvDXj
t:<sip:0893540105@sip>
i:dd4d34b9-df92-1239-eabe-00163ea67143
CSeq:31559588 INVITE
m:<sip:mod_sofia@27.127.233.78:5060>
User-Agent:HV
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
k:timer,path,replaces
u:talk,hold,conference,refer
Privacy:none
c:application/sdp
Content-Disposition:session
l:270
X-sbcid:05e2326c-64f0-11eb-91a5-0da75cbe6c81
X-FS-Support:update_display,send_info
P-Asserted-Identity:<sip:0477770613@27.127.233.78>

v=0
o=FreeSWITCH 1612204699 1612204700 IN IP4 27.127.233.78
s=FreeSWITCH
c=IN IP4 27.127.233.78
t=0 0
m=audio 22062 RTP/AVP 8 0 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

Feb 2 2021 08:46:01.959 AWST: //97668/DB8785AF8BEB/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKvN07HZDSHSm4c
From: <sip:0477770613@27.127.233.78>;tag=SFBvamK9HvDXj
To: <sip:0893540105@sip>
Date: Tue, 02 Feb 2021 00:46:01 GMT
Call-ID: dd4d34b9-df92-1239-eabe-00163ea67143
CSeq: 31559588 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0


Feb 2 2021 08:46:01.967 AWST: //97669/DB8785AF8BEB/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:765617@10.1.200.200:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK10605393
From: <sip:0477770613@10.1.100.200>;tag=16242B88-12B2
To: <sip:765617@10.1.200.200>
Date: Tue, 02 Feb 2021 00:46:01 GMT
Call-ID: DB88BE98-642611EB-8BF1FE72-5C880660@10.1.100.200
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3683091887-1680216555-2347499122-1552418400
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1612226761
Contact: <sip:0477770613@10.1.100.200:5060>
Expires: 60
Allow-Events: telephone-event
Max-Forwards: 64
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 270

v=0
o=CiscoSystemsSIP-GW-UserAgent 710 3223 IN IP4 10.1.100.200
s=SIP Call
c=IN IP4 10.1.100.200
t=0 0
m=audio 31288 RTP/AVP 8 0 101
c=IN IP4 10.1.100.200
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Feb 2 2021 08:46:01.971 AWST: //97669/DB8785AF8BEB/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK10605393
From: <sip:0477770613@10.1.100.200>;tag=16242B88-12B2
To: <sip:765617@10.1.200.200>
Date: Tue, 02 Feb 2021 00:46:01 GMT
Call-ID: DB88BE98-642611EB-8BF1FE72-5C880660@10.1.100.200
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0


Feb 2 2021 08:46:02.027 AWST: //97669/DB8785AF8BEB/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK10605393
From: <sip:0477770613@10.1.100.200>;tag=16242B88-12B2
To: <sip:765617@10.1.200.200>;tag=25113686~7bae6435-a194-400a-8806-8ddaeecb994f-19053361
Date: Tue, 02 Feb 2021 00:46:01 GMT
Call-ID: DB88BE98-642611EB-8BF1FE72-5C880660@10.1.100.200
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Server: Cisco-CUCM11.5
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP;x-cisco-qos-tcl=true
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-ID: 6eb0681e00105000a000007686fd47c9;remote=63ea10d8e01b969ce22c87ab25113686
P-Asserted-Identity: "Interite Main Line" <sip:633@10.1.200.200>
Remote-Party-ID: "Interite Main Line" <sip:633@10.1.200.200>;party=called;screen=yes;privacy=off
Contact: <sip:765617@10.1.200.200:5060>;+u.sip!devicename.ccm.cisco.com="SEP007686FD47C9";video
Content-Length: 0


Feb 2 2021 08:46:02.031 AWST: //97668/DB8785AF8BEB/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKvN07HZDSHSm4c
From: <sip:0477770613@27.127.233.78>;tag=SFBvamK9HvDXj
To: <sip:0893540105@sip>;tag=16242BC8-562
Date: Tue, 02 Feb 2021 00:46:01 GMT
Call-ID: dd4d34b9-df92-1239-eabe-00163ea67143
CSeq: 31559588 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:765617@10.1.222.2:5060>
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0


Feb 2 2021 08:46:02.859 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:9001146812400691@203.54.247.182 SIP/2.0
Via: SIP/2.0/UDP 185.16.38.123:59380;branch=z9hG4bK17996747
Max-Forwards: 70
From: <sip:20@203.54.247.182>;tag=1260658450
To: <sip:9001146812400691@203.54.247.182>
Call-ID: 763226880-719772895-2031494762
CSeq: 1 INVITE
Contact: <sip:20@185.16.38.123:59380>
Content-Type: application/sdp
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH
User-Agent: Linksys-SPA942

v=0
o=20 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

Feb 2 2021 08:46:02.867 AWST: //97670/DC1176A78BF2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 185.16.38.123:59380;branch=z9hG4bK17996747
From: <sip:20@203.54.247.182>;tag=1260658450
To: <sip:9001146812400691@203.54.247.182>
Date: Tue, 02 Feb 2021 00:46:02 GMT
Call-ID: 763226880-719772895-2031494762
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0


Feb 2 2021 08:46:02.867 AWST: //97670/DC1176A78BF2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 185.16.38.123:59380;branch=z9hG4bK17996747
From: <sip:20@203.54.247.182>;tag=1260658450
To: <sip:9001146812400691@203.54.247.182>;tag=16242F10-26D
Date: Tue, 02 Feb 2021 00:46:02 GMT
Call-ID: 763226880-719772895-2031494762
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=21
Content-Length: 0


Feb 2 2021 08:46:02.967 AWST: //97670/DC1176A78BF2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 185.16.38.123:59380;branch=z9hG4bK17996747
From: <sip:20@203.54.247.182>;tag=1260658450
To: <sip:9001146812400691@203.54.247.182>;tag=16242F10-26D
Date: Tue, 02 Feb 2021 00:46:02 GMT
Call-ID: 763226880-719772895-2031494762
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=21
Content-Length: 0


Feb 2 2021 08:46:03.167 AWST: //97670/DC1176A78BF2/SIP/Msg/ccsipDisplayMsg:
Sent:
terminal monitor SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 185.16.38.123:59380;branch=z9hG4bK17996747
From: <sip:20@203.54.247.182>;tag=1260658450
To: <sip:9001146812400691@203.54.247.182>;tag=16242F10-26D
Date: Tue, 02 Feb 2021 00:46:02 GMT
Call-ID: 763226880-719772895-2031494762
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=21
Content-Length: 0


Feb 2 2021 08:46:03.567 AWST: //97670/DC1176A78BF2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 185.16.38.123:59380;branch=z9hG4bK17996747
From: <sip:20@203.54.247.182>;tag=1260658450
To: <sip:9001146812400691@203.54.247.182>;tag=16242F10-26D
Date: Tue, 02 Feb 2021 00:46:02 GMT
Call-ID: 763226880-719772895-2031494762
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=21
Content-Length: 0


Feb 2 2021 08:46:03.859 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:765617@10.1.222.2:5060 SIP/2.0
v:SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKvN07HZDSHSm4c
Route:<sip:765617@203.54.247.182:20704>
Max-Forwards:65
f:<sip:0477770613@27.127.233.78>;tag=SFBvamK9HvDXj
t:<sip:0893540105@sip>
i:dd4d34b9-df92-1239-eabe-00163ea67143
CSeq:31559588 CANCEL
Reason:Q.850;cause=16;text=""
l:0


Feb 2 2021 08:46:03.863 AWST: //97668/DB8785AF8BEB/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKvN07HZDSHSm4c
From: <sip:0477770613@27.127.233.78>;tag=SFBvamK9HvDXj
To: <sip:0893540105@sip>
Date: Tue, 02 Feb 2021 00:46:03 GMT
Call-ID: dd4d34b9-df92-1239-eabe-00163ea67143
CSeq: 31559588 CANCEL
Content-Length: 0


Feb 2 2021 08:46:03.863 AWST: //97669/DB8785AF8BEB/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:765617@10.1.200.200:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK10605393
From: <sip:0477770613@10.1.100.200>;tag=16242B88-12B2
To: <sip:765617@10.1.200.200>
Date: Tue, 02 Feb 2021 00:46:01 GMT
Call-ID: DB88BE98-642611EB-8BF1FE72-5C880660@10.1.100.200
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1612226763
Reason: Q.850;cause=16
Content-Length: 0


Feb 2 2021 08:46:03.863 AWST: //97668/DB8785AF8BEB/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKvN07HZDSHSm4c
From: <sip:0477770613@27.127.233.78>;tag=SFBvamK9HvDXj
To: <sip:0893540105@sip>;tag=16242BC8-562
Date: Tue, 02 Feb 2021 00:46:03 GMT
Call-ID: dd4d34b9-df92-1239-eabe-00163ea67143
CSeq: 31559588 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Reason: Q.850;cause=16
Content-Length: 0


Feb 2 2021 08:46:03.867 AWST: //97669/DB8785AF8BEB/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK10605393
From: <sip:0477770613@10.1.100.200>;tag=16242B88-12B2
To: <sip:765617@10.1.200.200>
Date: Tue, 02 Feb 2021 00:46:03 GMT
Call-ID: DB88BE98-642611EB-8BF1FE72-5C880660@10.1.100.200
Server: Cisco-CUCM11.5
CSeq: 101 CANCEL
Content-Length: 0


Feb 2 2021 08:46:03.867 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:765617@10.1.222.2:5060 SIP/2.0
v:SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKvN07HZDSHSm4c
Route:<sip:765617@203.54.247.182:20704>
Max-Forwards:65
f:<sip:0477770613@27.127.233.78>;tag=SFBvamK9HvDXj
t:<sip:0893540105@sip>;tag=16242BC8-562
i:dd4d34b9-df92-1239-eabe-00163ea67143
CSeq:31559588 ACK
l:0


Feb 2 2021 08:46:03.867 AWST: //97669/DB8785AF8BEB/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK10605393
From: <sip:0477770613@10.1.100.200>;tag=16242B88-12B2
To: <sip:765617@10.1.200.200>;tag=25113686~7bae6435-a194-400a-8806-8ddaeecb994f-19053361
Date: Tue, 02 Feb 2021 00:46:03 GMT
Call-ID: DB88BE98-642611EB-8BF1FE72-5C880660@10.1.100.200
CSeq: 101 INVITE
Allow-Events: presence
Server: Cisco-CUCM11.5
Content-Length: 0


Feb 2 2021 08:46:03.871 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent: no mon
ICI-AU-CNN-VRO-01#ACK sip:765617@10.1.200.200:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.100.200:5060;branch=z9hG4bK10605393
From: <sip:0477770613@10.1.100.200>;tag=16242B88-12B2
To: <sip:765617@10.1.200.200>;tag=25113686~7bae6435-a194-400a-8806-8ddaeecb994f-19053361
Date: Tue, 02 Feb 2021 00:46:01 GMT
Call-ID: DB88BE98-642611EB-8BF1FE72-5C880660@10.1.100.200
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

Hi,

 

I think you need to change your destination-pattern under dial-peer 110. X isn't a valid wildcard which can be used within a destination pattern.

 

Your config reads:

 

dial-peer voice 110 voip
description ## outgoing cucm01 ##
preference 1
destination-pattern 76XX17
session protocol sipv2
session target ipv4:10.1.200.200
voice-class codec 1
voice-class sip options-keepalive
dtmf-relay rtp-nte
no vad

 

Can you change that to:

 

dial-peer voice 110 voip
description ## outgoing cucm01 ##
preference 1
destination-pattern 76....
session protocol sipv2
session target ipv4:10.1.200.200
voice-class codec 1
voice-class sip options-keepalive
dtmf-relay rtp-nte
no vad

 

Hi Scott, thanks for you help. 

 

This dial-peer is a temporary one that I've created to send these "weird" incoming calls to my CUCM so I wouldn't lose any calls till I fix this. It will be removed when I get the right "called number" from PSTN.

 

Thanks

Try with this configuration

voice service voip
 sip
  sip-profiles inbound
!
voice class sip-profiles 20
 request INVITE sip-header T copy "sip:(.*)@" u01
 request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
!
dial-peer voice <dial peer number you use> voip
 description Inbound calls from PSTN
 voice-class sip profiles 20 inbound

This give this output with the SIP profile test tool, that looks like it would be what you search for when the input SDP is this, that's taken from the output from your debug.

Feb 2 2021 08:46:01.955 AWST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:765617@10.1.222.2:5060 SIP/2.0
v:SIP/2.0/UDP 27.127.233.78;rport;branch=z9hG4bKvN07HZDSHSm4c
Route:<sip:765617@203.54.247.182:20704>
Max-Forwards:65
f:<sip:0477770613@27.127.233.78>;tag=SFBvamK9HvDXj
t:<sip:0893540105@sip>
i:dd4d34b9-df92-1239-eabe-00163ea67143
CSeq:31559588 INVITE
m:<sip:mod_sofia@27.127.233.78:5060>
User-Agent:HV
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY
k:timer,path,replaces
u:talk,hold,conference,refer
Privacy:none
c:application/sdp
Content-Disposition:session
l:270
X-sbcid:05e2326c-64f0-11eb-91a5-0da75cbe6c81
X-FS-Support:update_display,send_info
P-Asserted-Identity:<sip:0477770613@27.127.233.78>

v=0
o=FreeSWITCH 1612204699 1612204700 IN IP4 27.127.233.78
s=FreeSWITCH
c=IN IP4 27.127.233.78
t=0 0
m=audio 22062 RTP/AVP 8 0 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20

Snag_155714d.png

The orange lines shows what has been modified with the SIP profile.

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