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ciscoMFNAO
Beginner

SIP/2.0 404 Not Found - CUCM_TO_CUBE_ITSP Provider

Hi all,

We are configuring a new SIP Trunk to our new ITSP . Topology looks like: 

CUCM---SIP TRUNK - CUBE - SIP TRUNK - ITSP

Debug shows the error SIP/2.0 404 Not Found.

Any ideas what can be causing this issue. Please note CUBE show run and debugs.

For testing We called from 9521 phone extension to 0917775245 PSTN Number (0 is stripped by CUBE  to be allowed to PSTN)

 

PLEASE note debugs SIP ERROR/CALL/MESSAGES, debug voice ccapi inout and  show run attached.

---------------------- DEBUGS----------------------------------
JPTLDAMAN-RTSIP-01#debug ccsip messages
SIP Call messages tracing is enabled
JPTLDAMAN-RTSIP-01#ter monitor
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
Feb  7 08:22:12.347: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0917775245@172.29.8.146:5060 SIP/2.0
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa5213e80a597
From: "Timoteo Test" <sip:9521@172.29.8.46>;tag=448640~1ae110bc-fd42-e324-02ec-760a5a7946df-29983909
To: <sip:0917775245@172.29.8.146>
Date: Wed, 07 Feb 2018 08:37:05 GMT
Call-ID: 12225400-a7a1bab1-a017-2e081dac@172.29.8.46
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:172.29.8.46:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED
Cisco-Guid: 0304239616-0000065536-0000000260-0772283820
Session-Expires:  1800
P-Asserted-Identity: "Timoteo Test" <sip:9521@172.29.8.46>
Remote-Party-ID: "Timoteo Test" <sip:9521@172.29.8.46>;party=calling;screen=yes;privacy=off
Contact: <sip:9521@172.29.8.46:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 390

v=0
o=CiscoSystemsCCM-SIP 448640 1 IN IP4 172.29.8.46
s=SIP Call
c=IN IP4 172.30.52.49
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24678 RTP/AVP 9 124 0 8 116 18 101
a=rtpmap:9 G722/8000
a=rtpmap:124 iSAC/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Feb  7 08:22:12.355: //1193/122254000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:917775245@66.110.114.133:5060 SIP/2.0
Via: SIP/2.0/UDP 10.130.2.166:5060;branch=z9hG4bK1A187B
Remote-Party-ID: "Timoteo Test" <sip:9521@10.130.2.166>;party=calling;screen=yes;privacy=off
From: "Timoteo Test" <sip:9521@10.130.2.166>;tag=425FEB0-1D66
To: <sip:917775245@66.110.114.133>
Date: Wed, 07 Feb 2018 08:22:12 GMT
Call-ID: D4E1C40D-B1611E8-8524B63D-1C0EF207@10.130.2.166
Supported: 1XX,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0304239616-0000065536-0000000260-0772283820
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1517991732
Contact: <sip:9521@10.130.2.166:5060>
Call-Info: <sip:10.130.2.166:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 352

v=0
o=CiscoSystemsSIP-GW-UserAgent 8739 9819 IN IP4 10.130.2.166
s=SIP Call
c=IN IP4 10.130.2.166
t=0 0
m=audio 22060 RTP/AVP 9 0 8 18 101
c=IN IP4 10.130.2.166
a=rtpmap:9 G722/8000
a=fmtp:9 bitrate=56
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Feb  7 08:22:12.355: //1192/122254000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa5213e80a597
From: "Timoteo Test" <sip:9521@172.29.8.46>;tag=448640~1ae110bc-fd42-e324-02ec-760a5a7946df-29983909
To: <sip:0917775245@172.29.8.146>
Date: Wed, 07 Feb 2018 08:22:12 GMT
Call-ID: 12225400-a7a1bab1-a017-2e081dac@172.29.8.46
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Feb  7 08:22:12.383: //1193/122254000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.130.2.166:5060;branch=z9hG4bK1A187B
Call-ID: D4E1C40D-B1611E8-8524B63D-1C0EF207@10.130.2.166
From: "Timoteo Test"<sip:9521@10.130.2.166>;tag=425FEB0-1D66
To: <sip:917775245@66.110.114.133>
CSeq: 101 INVITE
Content-Length: 0


Feb  7 08:22:12.387: //1193/122254000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.130.2.166:5060;branch=z9hG4bK1A187B
Call-ID: D4E1C40D-B1611E8-8524B63D-1C0EF207@10.130.2.166
From: "Timoteo Test"<sip:9521@10.130.2.166>;tag=425FEB0-1D66
To: <sip:917775245@66.110.114.133>;tag=e07ac79e
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0


Feb  7 08:22:12.387: //1192/122254000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa5213e80a597
From: "Timoteo Test" <sip:9521@172.29.8.46>;tag=448640~1ae110bc-fd42-e324-02ec-760a5a7946df-29983909
To: <sip:0917775245@172.29.8.146>;tag=425FED0-1A5B
Date: Wed, 07 Feb 2018 08:22:12 GMT
Call-ID: 12225400-a7a1bab1-a017-2e081dac@172.29.8.46
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=1
Content-Length: 0


Feb  7 08:22:12.387: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:917775245@66.110.114.133:5060 SIP/2.0
Via: SIP/2.0/UDP 10.130.2.166:5060;branch=z9hG4bK1A187B
From: "Timoteo Test" <sip:9521@10.130.2.166>;tag=425FEB0-1D66
To: <sip:917775245@66.110.114.133>;tag=e07ac79e
Date: Wed, 07 Feb 2018 08:22:12 GMT
Call-ID: D4E1C40D-B1611E8-8524B63D-1C0EF207@10.130.2.166
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


Feb  7 08:22:12.391: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0917775245@172.29.8.146:5060 SIP/2.0
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa5213e80a597
From: "Timoteo Test" <sip:9521@172.29.8.46>;tag=448640~1ae110bc-fd42-e324-02ec-760a5a7946df-29983909
To: <sip:0917775245@172.29.8.146>;tag=425FED0-1A5B
Date: Wed, 07 Feb 2018 08:37:05 GMT
Call-ID: 12225400-a7a1bab1-a017-2e081dac@172.29.8.46
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

debug ccsip ?
  all        Enable all SIP debugging traces
  calls      Enable CCSIP SPI calls debugging trace
  dhcp       Enable SIP-DHCP debugging trace
  error      Enable SIP error debugging trace
  events     Enable SIP events debugging trace
  function   Enable SIP function debugging trace
  info       Enable SIP info debugging trace
  media      Enable SIP media debugging trace
  messages   Enable CCSIP SPI messages debugging trace
  preauth    Enable SIP preauth debugging traces
  states     Enable CCSIP SPI states debugging trace
  translate  Enable SIP translation debugging trace
  transport  Enable SIP transport debugging traces
  verbose    Enable verbose mode

JPTLDAMAN-RTSIP-01#debug ccsip
Feb  7 08:22:42.295: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.8.146:5060 SIP/2.0
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa5227d5d2e9
From: <sip:172.29.8.46>;tag=2083901419
To: <sip:172.29.8.146>
Date: Wed, 07 Feb 2018 08:37:35 GMT
Call-ID: 2403f700-a7a1bacf-a018-2e081dac@172.29.8.46
User-Agent: Cisco-CUCM10.5
CSeq: 101 OPTIONS
Contact: <sip:172.29.8.46:5060;transport=tcp>
Max-Forwards: 0
Content-Length: 0


Feb  7 08:22:42.299: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa5227d5d2e9
From: <sip:172.29.8.46>;tag=2083901419
To: <sip:172.29.8.146>;tag=42673A4-11A1
Date: Wed, 07 Feb 2018 08:22:42 GMT
Call-ID: 2403f700-a7a1bacf-a018-2e081dac@172.29.8.46
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: 1XX,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 450

v=0
o=CiscoSystemsSIP-GW-UserAgent 3932 5771 IN IP4 172.29.8.146
s=SIP Call
c=IN IP4 172.29.8.146
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.8.146
m=image 0 udptl t38
c=IN IP4 172.29.8.146
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy


-------------------------------------------------------------------------------------------------------


JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#debug ccsip error
SIP Call error tracing is enabled
JPTLDAMAN-RTSIP-01#no debug ccsip messages
SIP Call messages tracing is disabled
JPTLDAMAN-RTSIP-01#
SIP: (1195) Attribute mid, level 1 instance 1 not found.
Feb  7 08:23:11.963: //1195/354D03800000/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
SIP: (1195) Attribute ptime, level 1 instance 1 not found.
Feb  7 08:23:11.963: //1195/354D03800000/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
SIP: (1195) Attribute ptime, level 1 instance 1 not found.
Feb  7 08:23:11.963: //1195/354D03800000/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
SIP: (1195) Attribute ptime, level 1 instance 1 not found.
Feb  7 08:23:11.963: //1195/354D03800000/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
Feb  7 08:23:11.963: //1195/354D03800000/SIP/Error/sipSPI_ipip_update_call_entry:
failed to update call entry
Feb  7 08:23:11.963: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 1195
Feb  7 08:23:11.963: //1195/354D03800000/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.
Feb  7 08:23:11.967: //1196/354D03800000/SIP/Error/sip_iwf_sip_copy_channelInfo_to_sdp: We are either escalating, orno stream found for this m-line index:1
Feb  7 08:23:11.967: //1196/354D03800000/SIP/Error/sip_iwf_sip_copy_channelInfo_to_sdp: We are either escalating, orno stream found for this m-line index:1
Feb  7 08:23:11.967: //1196/354D03800000/SIP/Error/sip_iwf_sip_copy_channelInfo_to_sdp: We are either escalating, orno stream found for this m-line index:1
Feb  7 08:23:11.967: //1196/354D03800000/SIP/Error/sip_iwf_sip_copy_channelInfo_to_sdp: We are either escalating, orno stream found for this m-line index:1
SIP: (1196) Group (a= group line) attribute, level 65535 instance 1 not found.
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#no debug ccsip error
SIP Call error tracing is disabled
JPTLDAMAN-RTSIP-01#no debug ccsip ?
  all        Enable all SIP debugging traces
  calls      Enable CCSIP SPI calls debugging trace
  dhcp       Enable SIP-DHCP debugging trace
  error      Enable SIP error debugging trace
  events     Enable SIP events debugging trace
  function   Enable SIP function debugging trace
  info       Enable SIP info debugging trace
  media      Enable SIP media debugging trace
  messages   Enable CCSIP SPI messages debugging trace
  preauth    Enable SIP preauth debugging traces
  states     Enable CCSIP SPI states debugging trace
  translate  Enable SIP translation debugging trace
  transport  Enable SIP transport debugging traces
  verbose    Enable verbose mode

----------------------------------------------------------------------------------------------------------------------------


SIP Call statistics tracing is disabled
JPTLDAMAN-RTSIP-01#debug ccsip calls
SIP Call statistics tracing is enabled
JPTLDAMAN-RTSIP-01#
Feb  7 08:24:19.959: //1201/5DD4FD800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x313DBB08
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 9521
Called Number            : 917775245
Source IP Address (Sig  ): 10.130.2.166
Destn SIP Req Addr:Port  : 66.110.114.133:5060
Destn SIP Resp Addr:Port : 66.110.114.133:5060
Destination Name         : 66.110.114.133

Feb  7 08:24:19.959: //1201/5DD4FD800000/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): 10.130.2.166
Source IP Port    (Media): 23958
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

Feb  7 08:24:19.959: //1201/5DD4FD800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 1
Disconnect Cause (SIP)   : 404

Feb  7 08:24:19.967: //1200/5DD4FD800000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x313C5348
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 9521
Called Number            : 0917775245
Source IP Address (Sig  ): 172.29.8.146
Destn SIP Req Addr:Port  : 172.29.8.46:5060
Destn SIP Resp Addr:Port : 172.29.8.46:34093
Destination Name         : 172.29.8.46

Feb  7 08:24:19.967: //1200/5DD4FD800000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g722-64
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 9 (tx), 9 (rx)
Negotiated Dtmf-relay    : 8
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 172.29.8.146
Source IP Port    (Media): 28062
Destn  IP Address (Media): 172.30.52.49
Destn  IP Port    (Media): 24684
Orig Destn IP Address:Port (Media): [ - ]:0

Feb  7 08:24:19.967: //1200/5DD4FD800000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 1
Disconnect Cause (SIP)   : 404

JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#no debug ccsip calls
SIP Call statistics tracing is disabled
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#

------------------------------------------------------------------------------------------------------


JPTLDAMAN-RTSIP-01#debug voice ccapi inout
voip ccapi inout debugging is on
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
Feb  7 08:26:28.991: //-1/AAB8D4000000/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=9521
   ----- ccCallInfo IE subfields -----
   cisco-ani=9521
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=0917775245
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0

Feb  7 08:26:28.991: //-1/AAB8D4000000/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x2B6B74EC, Call Info(
   Calling Number=9521,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0917775245(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=1204
Feb  7 08:26:28.991: //-1/AAB8D4000000/CCAPI/ccCheckClipClir:
   In: Calling Number=9521(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
Feb  7 08:26:28.991: //-1/AAB8D4000000/CCAPI/ccCheckClipClir:
   Out: Calling Number=9521(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
Feb  7 08:26:28.991: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Feb  7 08:26:28.991: :cc_get_feature_vsa malloc success
Feb  7 08:26:28.991: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Feb  7 08:26:28.991:  cc_get_feature_vsa count is 1
Feb  7 08:26:28.991: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Feb  7 08:26:28.991: :FEATURE_VSA attributes are: feature_name:0,feature_time:711191832,feature_id:61
Feb  7 08:26:28.991: //1204/AAB8D4000000/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=9521(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0917775245(TON=Unknown, NPI=Unknown))
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/cc_process_call_setup_ind:
   Event=0x30D51900
Feb  7 08:26:28.995: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 0917775245
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/ccCallSetContext:
   Context=0x3153C1E8
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 1204 with tag 1 to app "_ManagedAppProcess_Default"
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=15, Params=0x31527638, Progress Indication=NULL(0)
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/ccCheckClipClir:
   In: Calling Number=9521(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/ccCheckClipClir:
   Out: Calling Number=9521(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/ccCallSetupRequest:
   Destination Pattern=^09........, Called Number=917775245, Digit Strip=FALSE
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/ccCallSetupRequest:
   Calling Number=9521(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=917775245(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=Timoteo Test
   Account Number=9521, Final Destination Flag=TRUE,
   Guid=AAB8D400-0001-0000-0000-010A2E081DAC, Outgoing Dial-peer=15
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=9521
   ----- ccCallInfo IE subfields -----
   cisco-ani=9521
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=917775245
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0

Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x2B6B74EC, Interface Type=3, Destination=, Mode=0x0,
   Call Params(Calling Number=9521,(Calling Name=Timoteo Test)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=917775245(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=15, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
Feb  7 08:26:28.995: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Feb  7 08:26:28.995: :cc_get_feature_vsa malloc success
Feb  7 08:26:28.995: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Feb  7 08:26:28.995:  cc_get_feature_vsa count is 2
Feb  7 08:26:28.995: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

Feb  7 08:26:28.995: :FEATURE_VSA attributes are: feature_name:0,feature_time:711192056,feature_id:62
Feb  7 08:26:28.995: //1205/AAB8D4000000/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
Feb  7 08:26:28.995: //1205/AAB8D4000000/CCAPI/ccCallSetContext:
   Context=0x315275E8
Feb  7 08:26:28.995: //1204/AAB8D4000000/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=15
Feb  7 08:26:28.999: //1205/AAB8D4000000/CCAPI/cc_api_call_proceeding:
   Interface=0x2B6B74EC, Progress Indication=NULL(0)
Feb  7 08:26:29.023: //1205/AAB8D4000000/CCAPI/cc_api_call_disconnected:
   Cause Value=1, Interface=0x2B6B74EC, Call Id=1205
Feb  7 08:26:29.023: //1205/AAB8D4000000/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=1, Retry Count=0)
Feb  7 08:26:29.023: //1204/AAB8D4000000/CCAPI/ccCallReleaseResources:
   release reserved xcoding resource.
Feb  7 08:26:29.023: //1205/AAB8D4000000/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=1205
Feb  7 08:26:29.023: //1205/AAB8D4000000/CCAPI/ccCallDisconnect:
   Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=1)
Feb  7 08:26:29.023: //1205/AAB8D4000000/CCAPI/ccCallDisconnect:
   Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
Feb  7 08:26:29.023: //1205/AAB8D4000000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x2B6B74EC, Tag=0x0, Call Id=1205,
   Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
Feb  7 08:26:29.023: //1205/AAB8D4000000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Feb  7 08:26:29.023: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Feb  7 08:26:29.023: :cc_free_feature_vsa freeing 2A63EDF0
Feb  7 08:26:29.023: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Feb  7 08:26:29.023:  vsacount in free is 1
Feb  7 08:26:29.023: //1204/AAB8D4000000/CCAPI/ccCallDisconnect:
   Cause Value=1, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Feb  7 08:26:29.023: //1204/AAB8D4000000/CCAPI/ccCallDisconnect:
   Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1)
Feb  7 08:26:29.027: //1204/AAB8D4000000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x2B6B74EC, Tag=0x0, Call Id=1204,
   Call Entry(Disconnect Cause=1, Voice Class Cause Code=0, Retry Count=0)
Feb  7 08:26:29.027: //1204/AAB8D4000000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Feb  7 08:26:29.027: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Feb  7 08:26:29.027: :cc_free_feature_vsa freeing 2A63ED10
Feb  7 08:26:29.027: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

Feb  7 08:26:29.027:  vsacount in free is 0
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#u all
All possible debugging has been turned off
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#

2 ACCEPTED SOLUTIONS

Accepted Solutions

Hello,

Everything looks ok at your side.

Please work with provider and get this issue fixed nothing can be done from your side.

 

They are not able to route the call due to some unknown reason and sending "404 Not Found".

 

Regards,

Balram


Please rate if you find this helpful
Thank You
Balram

View solution in original post

Amigo,

 

this will remain a guessing game. the provider is sending a 404, so there is something they are not liking either the called number or calling number. I suggest you get in touch with them and let them tell you want they require, so you can send the appropriate number of digits.

Please remember to rate useful posts, by clicking on the stars below.

View solution in original post

11 REPLIES 11
Ahmed Khalefa
Beginner

Hello ,
it seems that the ITSP didn't find a route for the dialed number ..

Feb 7 08:22:12.387: //1193/122254000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.130.2.166:5060;branch=z9hG4bK1A187B
Call-ID: D4E1C40D-B1611E8-8524B63D-1C0EF207@10.130.2.166
From: "Timoteo Test"<sip:9521@10.130.2.166>;tag=425FEB0-1D66
To: <sip:917775245@66.110.114.133>;tag=e07ac79e
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0



So , basic questions :
- Are you sure this is a correct Dialed Digits ?
- Do you have valid subscription for such dialing ? ( like for instance the CUCM will give you similar message if your CSS doesn't have appropriate level ) ..

Try to contact them if possible to check the dialing plan , sometimes the ITSP needs certain prefixes ..

Thanks A lot,
Ahmed Salah

Hi Ahmed,

Thanks for your replay. Yes I´, dialing the correct number, I must be sending a number like 917 775245 without 0 to my provider. I called them my subscription they say my subscription is okay. I able able to get traffic when I call my SIP DID Number from the PSTN by I get another error from their side (internal server error 500) and calls are not send to internal CUCM extensions. I Changed my internal extension to the complete DID number without success.

 

Any other ideias on what maybe causing this issue PLZ???

---------------

Feb  7 09:59:05.222: //1317/5B86788A85D4/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 10.131.0.210:5060;branch=z9hG4bK7vjjjknnyybjkuknniueeujc7
From: <sip:222706000@10.131.0.210>;tag=44319f99
To: <sip:222679521@10.130.2.166;user=phone>;tag=47EA388-646
Date: Wed, 07 Feb 2018 09:59:05 GMT
Call-ID: SBCe9aa56d1b3839c00e25becf20160e07c@softx3000
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=127
Content-Length: 0

 

Hello, Get below debugs for failed call. debug voip ccapi inout debug ccsip message debug voip translation Make sure you enable all 3 debugs and make test call to collect the logs. share running config from the CUBE along with debugs requested. Regards, Balram

Please rate if you find this helpful
Thank You
Balram

Please note the debugs and show run attached.

Thanks.

Hello,

Everything looks ok at your side.

Please work with provider and get this issue fixed nothing can be done from your side.

 

They are not able to route the call due to some unknown reason and sending "404 Not Found".

 

Regards,

Balram


Please rate if you find this helpful
Thank You
Balram

View solution in original post

Thanks for your support Balram,

I call them and schedule a troubleshoot session and figure out whats wrong.

Cheers,

Hello,

Looks like you are sending incorrect calling or called number to provider.
INVITE Received from CUCM:
Received:
INVITE sip:0917775245@172.29.8.146:5060 SIP/2.0
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa5213e80a597
From: "Timoteo Test" <sip:9521@172.29.8.46>;tag=448640~1ae110bc-fd42-e324-02ec-760a5a7946df-29983909
To: <sip:0917775245@172.29.8.146>

It has 0 in called number but when you sent it to provider 0 get stripped.

Sent:
INVITE sip:917775245@66.110.114.133:5060 SIP/2.0
Via: SIP/2.0/UDP 10.130.2.166:5060;branch=z9hG4bK1A187B
Remote-Party-ID: "Timoteo Test" <sip:9521@10.130.2.166>;party=calling;screen=yes;privacy=off
From: "Timoteo Test" <sip:9521@10.130.2.166>;tag=425FEB0-1D66
To: <sip:917775245@66.110.114.133>

And also you are sending calling number as 4 digit extension and i guess that what provider dislike and disconnect the call.

Could you please trying sending calling number as complete DID number and see if that works ?

Also ask you provider if they need 0 as prefix in called number.

Regards,
Balram

Please rate if you find this helpful
Thank You
Balram

Hi Balram,

Thanks for your replay. I´m sending the correct number. They acept a number like 917 7775245 without the 0. I try changing to the complete DID but without sucesss, please note debug bellow:

Any other ideas PLZ?

 

-------------

JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#debug ccsip messages
SIP Call messages tracing is enabled
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
Feb  7 09:48:58.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0917775245@172.29.8.146:5060 SIP/2.0
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa9ea196caf3f
From: "Timoteo Test" <sip:222679521@172.29.8.46>;tag=453899~1ae110bc-fd42-e324-02ec-760a5a7946df-29984105
To: <sip:0917775245@172.29.8.146>
Date: Wed, 07 Feb 2018 10:03:50 GMT
Call-ID: 308e4c80-a7a1cf06-a479-2e081dac@172.29.8.46
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:172.29.8.46:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=VIDEO_UNSPECIFIED
Cisco-Guid: 0814632064-0000065536-0000000284-0772283820
Session-Expires:  1800
P-Asserted-Identity: "Timoteo Test" <sip:222679521@172.29.8.46>
Remote-Party-ID: "Timoteo Test" <sip:222679521@172.29.8.46>;party=calling;screen=yes;privacy=off
Contact: <sip:222679521@172.29.8.46:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 390

v=0
o=CiscoSystemsCCM-SIP 453899 1 IN IP4 172.29.8.46
s=SIP Call
c=IN IP4 172.30.52.49
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24700 RTP/AVP 9 124 0 8 116 18 101
a=rtpmap:9 G722/8000
a=rtpmap:124 iSAC/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

SIP: (1299) Attribute mid, level 1 instance 1 not found.
Feb  7 09:48:58.166: //1299/308E4C800000/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
SIP: (1299) Attribute ptime, level 1 instance 1 not found.
Feb  7 09:48:58.166: //1299/308E4C800000/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
SIP: (1299) Attribute ptime, level 1 instance 1 not found.
Feb  7 09:48:58.166: //1299/308E4C800000/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
SIP: (1299) Attribute ptime, level 1 instance 1 not found.
Feb  7 09:48:58.166: //1299/308E4C800000/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
Feb  7 09:48:58.166: //1299/308E4C800000/SIP/Error/sipSPI_ipip_update_call_entry:
failed to update call entry
Feb  7 09:48:58.166: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 1299
Feb  7 09:48:58.166: //1299/308E4C800000/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.
Feb  7 09:48:58.166: //1299/308E4C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa9ea196caf3f
From: "Timoteo Test" <sip:222679521@172.29.8.46>;tag=453899~1ae110bc-fd42-e324-02ec-760a5a7946df-29984105
To: <sip:0917775245@172.29.8.146>
Date: Wed, 07 Feb 2018 09:48:58 GMT
Call-ID: 308e4c80-a7a1cf06-a479-2e081dac@172.29.8.46
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


Feb  7 09:48:58.170: //1300/308E4C800000/SIP/Error/sip_iwf_sip_copy_channelInfo_to_sdp: We are either escalating, orno stream found for this m-line index:1
Feb  7 09:48:58.170: //1300/308E4C800000/SIP/Error/sip_iwf_sip_copy_channelInfo_to_sdp: We are either escalating, orno stream found for this m-line index:1
Feb  7 09:48:58.170: //1300/308E4C800000/SIP/Error/sip_iwf_sip_copy_channelInfo_to_sdp: We are either escalating, orno stream found for this m-line index:1
Feb  7 09:48:58.170: //1300/308E4C800000/SIP/Error/sip_iwf_sip_copy_channelInfo_to_sdp: We are either escalating, orno stream found for this m-line index:1
SIP: (1300) Group (a= group line) attribute, level 65535 instance 1 not found.
Feb  7 09:48:58.174: //1300/308E4C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:917775245@66.110.114.133:5060 SIP/2.0
Via: SIP/2.0/UDP 10.130.2.166:5060;branch=z9hG4bK2516DC
Remote-Party-ID: "Timoteo Test" <sip:222679521@10.130.2.166>;party=calling;screen=yes;privacy=off
From: "Timoteo Test" <sip:222679521@10.130.2.166>;tag=4756DE8-B80
To: <sip:917775245@66.110.114.133>
Date: Wed, 07 Feb 2018 09:48:58 GMT
Call-ID: F3CA31FD-B2211E8-85BBB63D-1C0EF207@10.130.2.166
Supported: 1XX,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0814632064-0000065536-0000000284-0772283820
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1517996938
Contact: <sip:222679521@10.130.2.166:5060>
Call-Info: <sip:10.130.2.166:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 352

v=0
o=CiscoSystemsSIP-GW-UserAgent 9854 5052 IN IP4 10.130.2.166
s=SIP Call
c=IN IP4 10.130.2.166
t=0 0
m=audio 27548 RTP/AVP 9 0 8 18 101
c=IN IP4 10.130.2.166
a=rtpmap:9 G722/8000
a=fmtp:9 bitrate=56
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

Feb  7 09:48:58.198: //1300/308E4C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.130.2.166:5060;branch=z9hG4bK2516DC
Call-ID: F3CA31FD-B2211E8-85BBB63D-1C0EF207@10.130.2.166
From: "Timoteo Test"<sip:222679521@10.130.2.166>;tag=4756DE8-B80
To: <sip:917775245@66.110.114.133>
CSeq: 101 INVITE
Content-Length: 0


Feb  7 09:48:58.198: //1300/308E4C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.130.2.166:5060;branch=z9hG4bK2516DC
Call-ID: F3CA31FD-B2211E8-85BBB63D-1C0EF207@10.130.2.166
From: "Timoteo Test"<sip:222679521@10.130.2.166>;tag=4756DE8-B80
To: <sip:917775245@66.110.114.133>;tag=7aad131a
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0


Feb  7 09:48:58.202: //1299/308E4C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa9ea196caf3f
From: "Timoteo Test" <sip:222679521@172.29.8.46>;tag=453899~1ae110bc-fd42-e324-02ec-760a5a7946df-29984105
To: <sip:0917775245@172.29.8.146>;tag=4756E08-ECE
Date: Wed, 07 Feb 2018 09:48:58 GMT
Call-ID: 308e4c80-a7a1cf06-a479-2e081dac@172.29.8.46
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=1
Content-Length: 0


Feb  7 09:48:58.202: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:917775245@66.110.114.133:5060 SIP/2.0
Via: SIP/2.0/UDP 10.130.2.166:5060;branch=z9hG4bK2516DC
From: "Timoteo Test" <sip:222679521@10.130.2.166>;tag=4756DE8-B80
To: <sip:917775245@66.110.114.133>;tag=7aad131a
Date: Wed, 07 Feb 2018 09:48:58 GMT
Call-ID: F3CA31FD-B2211E8-85BBB63D-1C0EF207@10.130.2.166
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


Feb  7 09:48:58.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0917775245@172.29.8.146:5060 SIP/2.0
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa9ea196caf3f
From: "Timoteo Test" <sip:222679521@172.29.8.46>;tag=453899~1ae110bc-fd42-e324-02ec-760a5a7946df-29984105
To: <sip:0917775245@172.29.8.146>;tag=4756E08-ECE
Date: Wed, 07 Feb 2018 10:03:50 GMT
Call-ID: 308e4c80-a7a1cf06-a479-2e081dac@172.29.8.46
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0


Feb  7 09:49:05.674: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.8.146:5060 SIP/2.0
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa9eb87d22bc
From: <sip:172.29.8.46>;tag=1447228482
To: <sip:172.29.8.146>
Date: Wed, 07 Feb 2018 10:03:58 GMT
Call-ID: 35530080-a7a1cf0e-a47a-2e081dac@172.29.8.46
User-Agent: Cisco-CUCM10.5
CSeq: 101 OPTIONS
Contact: <sip:172.29.8.46:5060;transport=tcp>
Max-Forwards: 0
Content-Length: 0


Feb  7 09:49:05.678: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.29.8.46:5060;branch=z9hG4bKa9eb87d22bc
From: <sip:172.29.8.46>;tag=1447228482
To: <sip:172.29.8.146>;tag=4758B3C-105D
Date: Wed, 07 Feb 2018 09:49:05 GMT
Call-ID: 35530080-a7a1cf0e-a47a-2e081dac@172.29.8.46
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: 1XX,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 450

v=0
o=CiscoSystemsSIP-GW-UserAgent 6534 4327 IN IP4 172.29.8.146
s=SIP Call
c=IN IP4 172.29.8.146
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.8.146
m=image 0 udptl t38
c=IN IP4 172.29.8.146
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy

Amigo,

 

this will remain a guessing game. the provider is sending a 404, so there is something they are not liking either the called number or calling number. I suggest you get in touch with them and let them tell you want they require, so you can send the appropriate number of digits.

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View solution in original post

You´re right Dennis,
It´s very strange because we have another SIP TRunk with the same provider and they´re accepting the Called/Calling format the sending here. I schedule a troubleshooting session with them so we can investigate what´s really happening.

 

Thanks so much for your support.

 

 

Amigo,

 

this will remain a guessing game. the provider is sending a 404, so there is something they are not liking either the called number or calling number. I suggest you get in touch with them and let them tell you want they require, so you can send the appropriate number of digits.

Please remember to rate useful posts, by clicking on the stars below.

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