Inbound SIP cube call not working.

[devraj123]

Hi All,

When a particular customer phones the company we cannot pick up the call. Sometimes the customer receives a disconnection message.

When a particular customer phones the company we cannot pick up the call.Sometimes the customer receives a disconnection message.
Topology : They have 2 CUBE routers in site 1 3945 and site 2 1004 ASR. Site 1 works. but site 2 not.

there is no issue with outbound call.

After going through the new logs, my understanding is that, the same call flow works when using  platform ISRG2 3945e, but does not work with Platform ASR 1004. IOS are same, IOS-15.2.4.S5

fHi all,

When a particular customer phones the company we cannot pick up the call. Sometimes the customer receives a disconnection message.

When a particular customer phones the company we cannot pick up the call.Sometimes the customer receives a disconnection message.
Topology : They have 2 CUBE routers in site 1 3945 and site 2 1004 ASR. Site 1 works. but site 2 not.

there is no issue with outbound call.

After going through the new logs, my understanding is that, the same call flow works when using  platform ISRG2 3945e, but does not work with Platform ASR 1004. IOS are same, IOS-15.2.4.S5

for working call on Cube from same number-

Hi all,

When a particular customer phones the company we cannot pick up the call. Sometimes the customer receives a disconnection message.

When a particular customer phones the company we cannot pick up the call.Sometimes the customer receives a disconnection message.
Topology : They have 2 CUBE routers in site 1 3945 and site 2 1004 ASR. Site 1 works. but site 2 not.

there is no issue with outbound call.

After going through the new logs, my understanding is that, the same call flow works when using  platform ISRG2 3945e, but does not work with Platform ASR 1004. IOS are same, IOS-15.2.4.S5

for working call on Cube from same number-

##### Working call to from 074xx59666 to 2xx322301 via site 1 platform 3945
####, CUBE received INVITE with no DTMF. Doing inband voice

342555: Nov  5 13:28:29: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:2xx322301@10.2.20.1:5060 SIP/2.0
Via: SIP/2.0/UDP 203.x.x.x:5060;branch=z9hG4bK2jcasd101g11q191b700.1
From: "074xx59666 Easy Cut"<sip:747756855@10.83.154.138;user=phone>;tag=1913124215-1446694109354-
To: "2xx322301 2xx322301"<sip:2xx322301@xyz.com>
Call-ID: BW1428293540511151313262887@10.83.154.138
CSeq: 718807126 INVITE
Contact: <sip:747756855@203.x.x.x:5060;transport=udp>
Supported: 100rel
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,multipart/mixed
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 205
v=0
o=BroadWorks 2172449485 1 IN IP4 203.x.x.x
s=-
c=IN IP4203.x.x.x
t=0 0
m=audio 19768 RTP/AVP 8 0 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no

##### DTMF relay set to inband-voice

342705: Nov  5 13:28:29: //150879/1DF6F5DBB437/SIP/Media/sipSPIUpdCallWithSdpInfo:
          Stream type            : voice-only
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : 150879
          Negotiated Codec       : g711alaw, bytes :160
          Nego. Codec payload    : 8 (tx), 8 (rx)
          Negotiated DTMF relay  : inband-voice
          Negotiated NTE payload : 0 (tx), 0 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [10.1.20.2]:0
          Media Dest Addr/Port   : [203.x.x.x]:19768

##### Inbound dial-peer 6000
342741: Nov  5 13:28:29: //-1/1DF6F5DBB437/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x25D59B4C, Call Info(
   Calling Number=747756855,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=2xx322301(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=6000, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=150879

dial-peer voice 6000 voip
description Incoming calls from ITSP
 translation-profile incoming SIP-incoming
media flow-around
rtp payload-type nse 99
session protocol sipv2
session target sip-server
incoming called-number ^2……..
voice-class codec 100  
 dtmf-relay rtp-nte
fax rate disable
fax protocol pass-through g711alaw

##### Outbound dial-peer 7002 (DTMF relay configured is rtp-nte)

342782: Nov  5 13:28:29: //150879/1DF6F5DBB437/CCAPI/ccCallSetupRequest:
   Calling Number=00747756855(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=2301(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=074xx59666 Easy Cut
   Account Number=747756855, Final Destination Flag=TRUE,
   Guid=1DF6F5DB-82A4-11E5-B437-FB1E7C5ED846, Outgoing Dial-peer=7002

dial-peer voice 7002 voip
description Calls to test extensions via CUCM Subscriber
huntstop
preference 1
destination-pattern ^[2-4]...$
media flow-around
session protocol sipv2
session target ipv4:172.25.250.2
incoming called-number ^0T
voice-class codec 100  
 no voice-class sip outbound-proxy   
 dtmf-relay rtp-nte
fax rate disable
fax protocol pass-through g711alaw
ip qos dscp cs3 signaling
no vad

##### Sent INVITE to CUCM with DTMF relay method rtp-nte

343146: Nov  5 13:28:29: //150880/1DF6F5DBB437/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2301@172.25.250.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.20.1:5060;branch=z9hG4bK5FA01F04
Remote-Party-ID: "074xx59666 Easy Cut" <sip:00747756855@10.2.20.1>;party=calling;screen=no;privacy=off
From: "074xx59666 Easy Cut" <sip:00747756855@xyz.com>;tag=C5702570-1C40
To: <sip:2301@172.25.250.2>
Date: Thu, 05 Nov 2015 03:28:29 GMT
Call-ID: 1DF7E017-82A411E5-B43DFB1E-7C5ED846@10.2.20.1
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 0502724059-2191790565-3023567646-2086590534
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M3
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1446694109
Contact: <sip:00747756855@10.2.20.1:5060>
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 28
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 364

v=0
o=CiscoSystemsSIP-GW-UserAgent 4671 3164 IN IP4 10.2.20.1
s=SIP Call
c=IN IP4203.x.x.x
t=0 0
m=audio 19768 RTP/AVP 8 0 18 100 101
c=IN IP4203.x.x.x
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

##### Received 200 OK from the CUCM side with codec g711ulaw and DTMF method set to rtp-nte

Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.20.1:5060;branch=z9hG4bK5FA01F04
From: "074xx59666 Easy Cut" <sip:00747756855@xyz.com>;tag=C5702570-1C40
          
To: <sip:2301@172.25.250.2>;tag=90953~9fa5af45-a14a-4c92-a3e2-4726a239bbc1-60501861
Date: Thu, 05 Nov 2015 03:28:29 GMT
Call-ID: 1DF7E017-82A411E5-B43DFB1E-7C5ED846@10.2.20.1
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence, kpml
Supported: replaces
Server: Cisco-CUCM10.5
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uas
Require:  timer
P-Asserted-Identity: abc <sip:2301@172.25.250.2>
Remote-Party-ID: abc <sip:2301@172.25.250.2>;party=called;screen=yes;privacy=off
Contact: <sip:2301@172.25.250.2:5060>
Content-Type: application/sdp
Content-Length: 246

v=0
o=CiscoSystemsCCM-SIP 90953 1 IN IP4 172.25.250.2
s=SIP Call
c=IN IP4 172.25.39.191
b=TIAS:64000
b=CT:64
b=AS:64
t=0 0
m=audio 24812 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

##### CUBE to CUCM leg doing RTP-nTE with payload type 101
345321: Nov  5 13:28:33: //150880/1DF6F5DBB437/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 203.x.x.x
Source IP Port    (Media): 19768
Destn  IP Address (Media): 172.25.39.191
Destn  IP Port    (Media): 24812
Orig Destn IP Address:Port (Media): [ - ]:0

##### CUBE forwards 200 OK with the DTMF-relay set to none or inband to the ITSP
345523: Nov  5 13:28:33: //150879/1DF6F5DBB437/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.x.x.x:5060;branch=z9hG4bK2jcasd101g11q191b700.1
From: "074xx59666 Easy Cut"<sip:747756855@10.83.154.138;user=phone>;tag=1913124215-1446694109354-
To: "2xx322301 2xx322301"<sip:2xx322301@xyz.com>;tag=C57025B4-1566
Date: Thu, 05 Nov 2015 03:28:29 GMT
Call-ID: BW1428293540511151313262887@10.83.154.138
CSeq: 718807126 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: abc <sip:2301@10.2.20.1>;party=called;screen=yes;privacy=off
Contact: <sip:2xx322301@10.2.20.1:5060>
Supported: replaces
Server: Cisco-SIPGateway/IOS-15.3.3.M3
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 190

v=0
o=CiscoSystemsSIP-GW-UserAgent 9679 1508 IN IP4 10.2.20.1
s=SIP Call
c=IN IP4 172.25.39.191
t=0 0
m=audio 24812 RTP/AVP 0
c=IN IP4 172.25.39.191
a=rtpmap:0 PCMU/8000
a=ptime:20

##### CUBE to ITSP leg doing inband voice as can be seen below

345542: Nov  5 13:28:33: //150879/1DF6F5DBB437/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x26502CB0
State of The Call        : STATE_ACTIVE
TCP Sockets Used         : NO
Calling Number           : 747756855
Called Number            : 2xx322301
Source IP Address (Sig  ): 10.2.20.1
Destn SIP Req Addr:Port  : 203.x.x.x:5060
Destn SIP Resp Addr:Port : 203.x.x.x:5060
Destination Name         : 203.x.x.x

345543: Nov  5 13:28:33: //150879/1DF6F5DBB437/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 172.25.39.191
Source IP Port    (Media): 24812
Destn  IP Address (Media): 203.x.x.x
Destn  IP Port    (Media): 19768
Orig Destn IP Address:Port (Media): [ - ]:0

+++++++++++++++++++++++++++++++++++++++++++++++++


###### Now coming to the non-working call from 074xx59666  to 738772301 via platform ASR 1004
###### Cube received INVITE from ITSP with no DTMF method configured

1208793: Nov  5 13:33:39.592 AEST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:738772301@10.2.10.1:5060 SIP/2.0
Via: SIP/2.0/UDP 203.x.x.x:5060;branch=z9hG4bKl7dvg800b8v1g0ti7pj0.1
From: "074xx59666 Easy Cut"<sip:747756855@203.x.x.x:5060;user=phone;url-cookie=VNCHHA1-1uu40pfqll9a2>;tag=764154492-1446694419570-
To: "738772301 738772301"<sip:738772301@xyz.com;url-cookie=VNCHHA1-jt3a25k0o4e7b>
Call-ID: BW14333957005111530146170@10.83.154.138
CSeq: 718962234 INVITE
Contact: <sip:747756855@203.x.x.x:5060;url-cookie=VNCHHA1-om4jh48tv6561;transport=udp>
Supported: 100rel
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,multipart/mixed
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 205

v=0
o=BroadWorks 2172624714 1 IN IP4 203.x.x.x
s=-
c=IN IP4 203.x.x.x
t=0 0
m=audio 19588 RTP/AVP 8 0 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no

###### Initially media is flow around

1208830: Nov  5 13:33:39.594 : //-1/D6E025DBB325/SIP/Info/sipSPIGetCallConfig: Media Antitrombone disabled
1208831: Nov  5 13:33:39.594 : //-1/D6E025DBB325/SIP/Info/sipSPISetMediaFlowMode: Storing the configured mode as FLOW-AROUND
1208832: Nov  5 13:33:39.594 : //-1/D6E025DBB325/SIP/Info/sipSPISetMediaFlowMode: xcoder high-density disabled
1208833: Nov  5 13:33:39.594 : //-1/D6E025DBB325/SIP/Info/sipSPISetMediaFlowMode: Flow Mode set to FLOW_AROUND

###### Incoming dial-peer 6000

1208989: Nov  5 13:33:39.600 : //-1/D6E025DBB325/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x3CBEDD24, Call Info(
   Calling Number=747756855,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=738772301(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=6000, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=3977694

dial-peer voice 6000 voip
description Incoming calls from ITSP
translation-profile incoming SIP-incoming-
media flow-around
session protocol sipv2
session target sip-server
incoming called-number ^7........
voice-class codec 100  
 dtmf-relay rtp-nte
fax rate disable
fax protocol pass-through g711alaw
ip qos dscp cs3 signaling
no vad

###### Outgoing dial-peer 7000

1209026: Nov  5 13:33:39.605 : //3977694/D6E025DBB325/CCAPI/ccCallSetupRequest:
   Calling Number=00747756855(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=0738772301(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=074xx59666 Easy Cut
   Account Number=747756855, Final Destination Flag=TRUE,
   Guid=D6E025DB-82A4-11E5-B325-D4B467A2F62F, Outgoing Dial-peer=7000

dial-peer voice 7000 voip
description Calls to and from extensions via CUCM Subscriber for ITSP
translation-profile outgoing 10DNIS
preference 1
destination-pattern ^18..........$
media flow-around
session protocol sipv2
session target ipv4:172.25.33.2
incoming called-number ^0T
voice-class codec 100  
 no voice-class sip outbound-proxy   
 dtmf-relay rtp-nte
fax rate disable
fax protocol pass-through g711alaw
ip qos dscp cs3 signaling
no vad

##### Require transcoder for DTMF mismatch

1209170: Nov  5 13:33:39.609 : //3977695/D6E025DBB325/SIP/Info/sipSPIDtmfTranscoder: local DTMF 6, peer DTMF 0
1209171: Nov  5 13:33:39.609 AEST: //3977695/D6E025DBB325/SIP/Info/sipSPIDtmfTranscoder: local codec 6, peer codec 6
1209172: Nov  5 13:33:39.610 AEST: //3977695/D6E025DBB325/SIP/Info/sipSPICheckAndReserveTranscoder: need transcoding for dtmf mismatch
1209173: Nov  5 13:33:39.610 AEST: //3977695/D6E025DBB325/SIP/Info/sipSPISrtpInterworking:

##### Mode changed to pass-through

1209200: Nov  5 13:33:39.610 AEST: //3977695/D6E025DBB325/SIP/Info/sipSPISetFlowModeBeforeXcoderReservation: Revert from FA to FT
1209201: Nov  5 13:33:39.610 AEST: //3977695/D6E025DBB325/SIP/Media/sipSPI_ipip_process_xcoder_flowthru_reversion: Revert to flow-through
1209202: Nov  5 13:33:39.610 AEST: //3977695/D6E025DBB325/SIP/Info/sipSPI_ipip_process_xcoder_flowthru_reversion:

1209223: Nov  5 13:33:39.611 AEST: //3977694/D6E025DBB325/SIP/Info/sip_iwf_common_fa2ft_ed_flow_mode_hdlr:
1209224: Nov  5 13:33:39.611 AEST: //3977694/D6E025DBB325/SIP/Info/sip_iwf_common_fa2ft_ed_flow_mode_hdlr: Changing the media flow mode to FLOW-THROUGH
1209225: Nov  5 13:33:39.611 AEST: //3977694/D6E025DBB325/SIP/Media/sipSPI_ipip_process_xcoder_flowthru_reversion: Revert to flow-through

##### Media inactivity timer started

1209282: Nov  5 13:33:39.613 : voip_rtp_set_non_rtp_call: Non-RTP call end. data-call-type=0
1209283: Nov  5 13:33:39.613 : voip_rtp_get_gccb:Error, invalid callID:-1
1209284: Nov  5 13:33:39.613 : //3977694/D6E025DBB325/SIP/Info/sipSPICreateAndStartRtpTimer: Valid RTP/RTCP session found and CLI enabled to create and start the inactivity timer.

##### Can see media inactivity criteria configured only on ASR C617rt14 but not on ISR C612rt1

gateway
 media-inactivity-criteria all
timer receive-rtcp 720
timer receive-rtp 1200

##### After that we see the Transcoder reservation failing so reverts back to flow around but I believe the media inactivity is still on

1209319: Nov  5 13:33:39.614 : //3977695/D6E025DBB325/SIP/Info/ccsip_get_xcode_resource: cap1=7, cap2=7
1209320: Nov  5 13:33:39.614 : //3977695/D6E025DBB325/SIP/Info/sipSPICheckAndReserveTranscoder: Xcoder reservation failed for dtmf. Dont disconnect the call.
1209321: Nov  5 13:33:39.614 : //3977695/D6E025DBB325/SIP/Info/sipSPI_ipip_vcc_ResetXcoder: Post to state machine..
1209322: Nov  5 13:33:39.614 : //3977695/D6E025DBB325/SIP/Info/ccsip_ipip_media_service_get_event_data: Event id = 17
1209323: Nov  5 13:33:39.614 : //3977695/D6E025DBB325/SIP/Info/ccsip_ipip_media_service_process_event: IPIP media service in use, deferring event
1209324: Nov  5 13:33:39.614 : //3977695/D6E025DBB325/SIP/Info/sipSPI_ipip_vcc_ResetXcoder: Posting event 188 to peer leg

##### Sent INVITE to CUCM with rtp-nte 101

1209505: Nov  5 13:33:39.623 AEST: //3977695/D6E025DBB325/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0738772301@172.25.33.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.10.1:5060;branch=z9hG4bK207C91A0B
Remote-Party-ID: "074xx59666 Easy Cut" <sip:00747756855@10.2.10.1>;party=calling;screen=no;privacy=off
From: "074xx59666 Easy Cut" <sip:00747756855@xyz.com>;tag=C59B3554-F72
To: <sip:0738772301@172.25.33.2>
Call-ID: D6E3F6BA-82A411E5-B32BD4B4-67A2F62F@10.2.10.1
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3605013979-2191790565-3005600948-1738733103
User-Agent: Cisco-SIPGateway/IOS-15.2.4.S5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1446694419
Contact: <sip:00747756855@10.2.10.1:5060>
Expires: 300
Allow-Events: telephone-event
Max-Forwards: 28
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 303

v=0
o=CiscoSystemsSIP-GW-UserAgent 6043 7143 IN IP4 10.2.10.1
s=SIP Call
c=IN IP4 203.x.x.x
t=0 0
m=audio 19588 RTP/AVP 8 0 18 101
c=IN IP4 203.x.x.x
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

###### We received 200 OK from the CUCM for the call with DTMF type rtp-nte payload 101

1209661: Nov  5 13:33:42.671 AEST: //3977695/D6E025DBB325/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.10.1:5060;branch=z9hG4bK207C91A0B
From: "074xx59666 Easy Cut" <sip:00747756855@xyz.com>;tag=C59B3554-F72
To: <sip:0738772301@172.25.33.2>;tag=1879916~9fa5af45-a14a-4c92-a3e2-4726a239bbc1-42301049
Call-ID: D6E3F6BA-82A411E5-B32BD4B4-67A2F62F@10.2.10.1
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence
Supported: replaces
Server: Cisco-CUCM10.5
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires:  1800;refresher=uas
Require:  timer
P-Asserted-Identity: xxx <sip:2301@172.25.33.2>
          
Remote-Party-ID: xxx <sip:2301@172.25.33.2>;party=called;screen=yes;privacy=off
Contact: <sip:0738772301@172.25.33.2:5060>
Content-Type: application/sdp
Content-Length: 247

v=0
o=CiscoSystemsCCM-SIP 1879916 1 IN IP4 172.25.33.2xxx
s=SIP Call
c=IN IP4 172.25.39.191
b=TIAS:64000
b=CT:64
b=AS:64
t=0 0
m=audio 22888 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

##### CUBE disconnects the call due to interworking error

1210011: Nov  5 13:33:42.683 AEST: //3977694/D6E025DBB325/CCAPI/ccConferenceCreate:
   Total Gain: Originating Party Direction=0dB, Terminating Party Direction=0dB
1210012: Nov  5 13:33:42.683 AEST: //3977694/D6E025DBB325/SIP/Info/ccsip_query_codec_info: Negotiated codec = 5
1210013: Nov  5 13:33:42.683 AEST: //3977694/D6E025DBB325/SIP/Info/sipSPI_ipip_codec_byte_transrating: codec class not supported in xrating scenario, return FALSE
1210014: Nov  5 13:33:42.683 AEST: voip_rtp_get_gccb:Error, no gccb for callID:3977695
1210015: Nov  5 13:33:42.684 AEST: //3977695/D6E025DBB325/SIP/Info/ccsip_call_statistics: Stats are not supported for IPIP call.
1210016: Nov  5 13:33:42.684 AEST: //3977695/D6E025DBB325/CCAPI/ccCallDisconnect:
   Cause Value=0, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)

##### Sent BYE to ITSP and 500 internal server error to CUCM leg after 3 seconds

1210065: Nov  5 13:33:42.687 xxx: //3977695/D6E025DBB325/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:0738772301@172.25.33.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.10.1:5060;branch=z9hG4bK207CB96E
From: "074xx59666 Easy Cut" <sip:00747756855@xyz.com>;tag=C59B3554-F72
To: <sip:0738772301@172.25.33.2>;tag=1879916~9fa5af45-a14a-4c92-a3e2-4726a239bbc1-42301049
Date: Thu, 05 Nov 2015 03:33:40 GMT
Call-ID: D6E3F6BA-82A411E5-B32BD4B4-67A2F62F@10.2.10.1
User-Agent: Cisco-SIPGateway/IOS-15.2.4.S5
Max-Forwards: 70
Timestamp: 1446694422
CSeq: 102 BYE
Reason: Q.850;cause=0
Content-Length: 0


1210066: Nov  5 13:33:42.687 xxx: //3977694/D6E025DBB325/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 203.x.x.x:5060;branch=z9hG4bKl7dvg800b8v1g0ti7pj0.1
From: "074xx59666 Easy Cut"<sip:747756855@203.x.x.x:5060;user=phone;url-cookie=VNCHHA1-1uu40pfqll9a2>;tag=764154492-1446694419570-
To: "738772301 738772301"<sip:738772301@xyz.com;url-cookie=VNCHHA1-jt3a25k0o4e7b>;tag=C59B3759-914
Call-ID: BW14333957005111530146170@10.83.154.138
CSeq: 718962234 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.S5
Reason: Q.850;cause=16
Content-Length: 0

++++++++++++++++++++++++++++++++


really appreciate your help on this....


Dev

[Ayodeji Okanlawon]

Dev,

Can we have the full debug ccsip all from the ASR please..Please configure the ff before you enable the logs.

service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit

Then..

<Enable debugs, then test again.>

debug ccsip all

<Enable session capture to txt file in terminal program.> (such as Putty)

then do the ff:

terminal length 0
show logging

Can you also inlcude the sh run of the ASR. May I ask why you are using flow around on the ASR? Is this by design?

NB: If you intend to mask the ips, please use a unique ip for each leg. Its important to see the full details. 203.X.X.X doesnt help because we dont know if the IP is for CUBE or ITSP..

Also I may not respond quickly as I am in and out of stuff today, but I will endeavour to lend a hand as much as I can

[devraj123]

Hi

version 15.2

dial-peer voice 6000 voip
 description Incoming calls from Telstra
 translation-profile incoming SIP-incoming-Telstra
 media flow-around
 rtp payload-type nse 99
 session protocol sipv2
 session target sip-server
 incoming called-number ^7........
 voice-class codec 100  
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 fax rate disable
 fax protocol pass-through g711alaw
 ip qos dscp cs3 signaling
 no vad
!

dial-peer voice 6002 voip
 description Outbound calls to Telstra
 translation-profile outgoing SIP-outgoing-Telstra
 destination-pattern ^0T
 media flow-around
 session protocol sipv2
 session target sip-server
 voice-class codec 100  
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 fax rate disable
 fax protocol pass-through g711alaw
 ip qos dscp cs3 signaling
 no vad
!



!
dial-peer voice 7000 voip
 description Calls to and from extensions via CUCM Subscriber for Telstra
 translation-profile outgoing 10DNIS
 preference 1
 destination-pattern ^18..........$
 media flow-around
 session protocol sipv2
 session target ipv4:172.25.33.2
 incoming called-number ^0T
 voice-class codec 100  
 no voice-class sip outbound-proxy   
 dtmf-relay rtp-nte
 fax rate disable
 fax protocol pass-through g711alaw
 ip qos dscp cs3 signaling
 no vad
!


gateway
 media-inactivity-criteria all
 timer receive-rtcp 720
 timer receive-rtp 1200
!
sip-ua
 set pstn-cause 47 sip-status 486
 retry invite 2
 retry response 3
 retry bye 3
 retry prack 6
 retry register 2
 timers expires 300000
 timers hold 45
 registrar dns:xxxx expires 3600
 sip-server dns:xxxxxx
 connection-reuse
 g729-annexb override
 

[Ayodeji Okanlawon]

Hi,

I looked at the logs and indeed its a veru strange one..The disconnect is coming from the cube and the disconnect cause is zero which means there is no error..Hence why you see the BYE sent to the ITSP as cause code 16 normal call clearing..

The key thing here is that the BYE comes immediately after the CUCM sends a 200 OK, the CUBE sends an ACK followed by a BYE. This usually suggests there is something in the 200 OK that the CUBE doesnt like. 

Looking at the SDP in the 200OK, I dont see anything obvious other than the fact that CUCM specifies a ptime in its SDP where as the INVITE from the ITSP doesnt have a ptime.

4278004: Nov 17 08:43:16.748 AEST: //5235687/419D3B589E19/SIP/Info/ccsip_query_codec_info: Negotiated codec = 5
4278005: Nov 17 08:43:16.748 AEST: //5235687/419D3B589E19/SIP/Info/sipSPI_ipip_codec_byte_transrating: codec class not supported in xrating scenario, return FALSE
4278006: Nov 17 08:43:16.748 AEST: voip_rtp_get_gccb:Error, no gccb for callID:5235688
4278007: Nov 17 08:43:16.748 AEST: //5235688/419D3B589E19/SIP/Info/ccsip_call_statistics: Stats are not supported for IPIP call.
4278008: Nov 17 08:43:16.748 AEST: //5235688/419D3B589E19/CCAPI/ccCallDisconnect:
   Cause Value=0, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
4278009: Nov 17 08:43:16.748 AEST: //5235688/419D3B589E19/CCAPI/ccCallDisconnect:
   Cause Value=0, Call Entry(Responsed=TRUE, Cause Value=0)

Can you send a debug ccsip all for a working call. We may be able to compare the capabilites..

[devraj123]

Hi Ayodeji,

######Look at the DTMF relay below for working call

below legs are coming from ITSP

1522245: Jun 25 09:07:42.651 AEST: //146398/A6DA0E808797/SIP/Media/sipSPIDisplayStreamInfo:
          Stream type            : voice+dtmf
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : 146398
          Peer Callid            : -1
          RTP/SRTP Negotiated     : 8
          Negotiated Codec       : g711alaw, bytes :160
          Nego. Codec payload    : 8 (tx), 8 (rx)
          Negotiated DTMF relay  : rtp-nte
          Negotiated NTE payload : 97 (tx), 97 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [203.3.3.3]:18100
          Media Dest Addr/Port   : [203.3.3.3]:18100

######Non-working call on

1506004: Jun 25 09:05:28.943 AEST: //146312/5727FC3186DA/SIP/Media/sipSPIDisplayStreamInfo:
          Stream type            : voice-only
          Media line             : 1
          State                  : STREAM_ADDING (2)
          Stream address type    : 1
          Callid                 : 146312
          Peer Callid            : -1
          RTP/SRTP Negotiated     : 8
          Negotiated Codec       : g711alaw, bytes :160
          Nego. Codec payload    : 8 (tx), 8 (rx)
          Negotiated DTMF relay  : inband-voice
          Negotiated NTE payload : 0 (tx), 0 (rx)
          Negotiated CN payload  : 0
          Media Srce Addr/Port   : [203.3.3.3]:18100
          Media Dest Addr/Port   : [203.3.3.3]:18100


###### That's because for the Working call we can see the telephony-event type as 97 for rtp-nte

[Ayodeji Okanlawon]

I understand what you are saying however CUBE should not fail a call because of DTMF issues. Normal call should work and only DTMF capabilites will be affected unless the ASR platform has bugs etc.

You can try and configure an on-board transcoder on the CUBE. This is because you need a xcoder to interwork between rtp-nte and inband voice

http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf-relay.html

As a lats resor if that fails, then I can only advise you to open a TAC case at this point because I dont know what else to suggest.

[Ayodeji Okanlawon]

To elininate your doubt about DTMF being the issue, try and configure an onboard xcoder on the CUBE. This is because CUBE can interwork between rtp-nte and inband voice using a xcoder.

You can also enable MTP on the sip trunk ( this is just to eliminate DTMF as the issue)

[sreekumara93]

Hello Ayodeji,

Is there any work around for this issue? I am having the same Problem, only a single external user unable to place the call to an internal number. The log also the same, Ptime is added by call manager in the 200 ok message.

The BYE message is coming from CUBE router with the cause value of 0.

Reason: Q.850;cause=0

Thanks,

Sreekumar

[avinsrid89]

Can I please see working debug ccsip all, debug voip ccapi inout  please?

Give calling and called number too or CALL-ID.

I believe this second RE-INVITE in the calling leg is wrong:

4277334: Nov 17 08:43:14.909 AEST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:738772301@10.2.91.1:5060 SIP/2.0
Via: SIP/2.0/UDP 203.52.0.0:5060;branch=z9hG4bKn2ml2i303olkspo20bv0.1
From: "0747259366 Easy Cut"<sip:747756855@203.52.0.0:5060;user=phone;url-cookie=VNCHHA1-1uu40pfqll9a2>;tag=223699450-1447713794080-
To: "738772301 738772301"<sip:738772301@xyz.com;url-cookie=VNCHHA1-468c3mtcqn6a1>
Call-ID: BW094314080171115-188344497@10.83.154.138
CSeq: 154907665 INVITE
Contact: <sip:747756855@203.52.0.0:5060;url-cookie=VNCHHA1-om4jh48tv6561;transport=udp>
Supported: 100rel
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,multipart/mixed
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 205

v=0
o=BroadWorks 2483766452 1 IN IP4 203.52.0.0
s=-
c=IN IP4 203.52.0.0
t=0 0
m=audio 19736 RTP/AVP 8 0 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no

~Avinash

[Ayodeji Okanlawon]

I am not sure that is a Re-INVITE. A RE-INVITE has a tag in the To header. This doesn't. I did see it but it's definitely not a re invite 

[Ayodeji Okanlawon]

Listen, I didn't ask or beg for a rating. Never rate my post with two stars if you deem it not good. Just leave it. We try our best here and personally I find it insulting to give such derogatory ratings 

[devraj123]

Hi Ayodeji,

I apologise if you feel.Sorry for that.

Deva

[avinsrid89]

The problem is indeed that INVITE I posted. It is not a RE-INVITE, hence it is wrong for ISP to send that second INVITE. TThe RE INVITE will come after session is established. Here the second INVITE comes before session establishment. This is the problem here. Hence here is what is happening:

1st INVITE:

4276045: Nov 17 08:43:14.409 AEST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:738772301@10.2.91.1:5060 SIP/2.0
Via: SIP/2.0/UDP 203.52.0.0:5060;branch=z9hG4bKn2ml2i303olkspo20bv0.1
From: "0747259366 Easy Cut"<sip:747756855@203.52.0.0:5060;user=phone;url-cookie=VNCHHA1-1uu40pfqll9a2>;tag=223699450-1447713794080-
To: "738772301 738772301"<sip:738772301@xyz.com;url-cookie=VNCHHA1-468c3mtcqn6a1>
Call-ID: BW094314080171115-188344497@10.83.154.138
CSeq: 154907665 INVITE
Contact: <sip:747756855@203.52.0.0:5060;url-cookie=VNCHHA1-om4jh48tv6561;transport=udp>
Supported: 100rel
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,multipart/mixed
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 205

v=0
o=BroadWorks 2483766452 1 IN IP4 203.52.0.0
s=-
c=IN IP4 203.52.0.0
t=0 0
m=audio 19736 RTP/AVP 8 0 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no

Everything goes well with this, after few message exchanges we get an INVITE again with same CALL-ID. This is not a REINVITE however:

4277334: Nov 17 08:43:14.909 AEST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:738772301@10.2.91.1:5060 SIP/2.0
Via: SIP/2.0/UDP 203.52.0.0:5060;branch=z9hG4bKn2ml2i303olkspo20bv0.1
From: "0747259366 Easy Cut"<sip:747756855@203.52.0.0:5060;user=phone;url-cookie=VNCHHA1-1uu40pfqll9a2>;tag=223699450-1447713794080-
To: "738772301 738772301"<sip:738772301@xyz.com;url-cookie=VNCHHA1-468c3mtcqn6a1>
Call-ID: BW094314080171115-188344497@10.83.154.138
CSeq: 154907665 INVITE
Contact: <sip:747756855@203.52.0.0:5060;url-cookie=VNCHHA1-om4jh48tv6561;transport=udp>
Supported: 100rel
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,multipart/mixed
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 205

v=0
o=BroadWorks 2483766452 1 IN IP4 203.52.0.0
s=-
c=IN IP4 203.52.0.0
t=0 0
m=audio 19736 RTP/AVP 8 0 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no

The session 1 is not even completed as we still did not get a proper 200 response for the first INVITE. 2nd INVITE is not looking right. CUBE does not like this INVITE, sends 500

4278060: Nov 17 08:43:16.751 AEST: //5235687/419D3B589E19/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 203.52.0.0:5060;branch=z9hG4bKn2ml2i303olkspo20bv0.1
From: "0747259366 Easy Cut"<sip:747756855@203.52.0.0:5060;user=phone;url-cookie=VNCHHA1-1uu40pfqll9a2>;tag=223699450-1447713794080-
To: "738772301 738772301"<sip:738772301@xyz.com;url-cookie=VNCHHA1-468c3mtcqn6a1>;tag=273787B-F6D
Date: Mon, 16 Nov 2015 22:43:14 GMT
Call-ID: BW094314080171115-188344497@10.83.154.138
CSeq: 154907665 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.S5
Reason: Q.850;cause=16
Content-Length: 0

This is ISP issue. Please contact your ISP to get this fixed.

Rate this post if you find it useful!

~Avinash

[devraj123]

Hi Ayodeji,

please find the debug logs for
debug ccsip all
debug ccapi inout

Thanks

Dev

[devraj123]

Hi Ayodeji,

This issue is with particular inbound customer call. this is not for everty inbound call.

Deva