Problem with external na call- CISCO 2801 -pbx

[acazarkov]

hello ,

I have configure cisco 2801 for Voip service.The network has 10 phones 6921 and they all were register . The problem is this: I can call and foreign and local numbers but I can not accept call from an external number. Which commands show and debbug can find where is the problem?

[paolo bevilacqua]

but I can not accept the invitation to accept an external number.

In simple words, what that means?

[acazarkov]

sorry, my misteke. I meen :I can not accept call from an external number

[paolo bevilacqua]

You did not even said which type of phone lines you have.

[acazarkov]

It is SIP TRUNK.

[paolo bevilacqua · Accepted solution]

You can simply add your PSTN number, as it appears in "debug ccsip message" with "term mon", as a secondary number on the ephone-dn that you want to receive external calls

Other call handling choices would require different configuration.

Please remember that when you expose a problem in a technical forum, you need to mention all the relevant facts.

Not anybody can understand that saying "the invitation to accept" you mean SIP INVITE on a SIP trunk.

[acazarkov]

this is my configuration :

voice service voip

ip address trusted list

  ipv4 10.0.0.18 255.255.255.255

gcid

callmonitor

no cti shutdown

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fall

h323

  call start slow

modem passthrough nse codec g711alaw

sip

  header-passing

  registrar server expires max 1200 min 60

  early-offer forced

  midcall-signaling passthru

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

voice translation-rule 1

rule 1 /201/ /38112719xxxx/

rule 2 /202/ /38112719xxxx/

rule 4 /204/ /38112719xxxx/

rule 5 /205/ /38112719xxxx/

rule 6 /206/ /38112719xxxx/

rule 7 /207/ /38112719xxxx/

rule 8 /208/ /38112719xxxx/

rule 9 /209/ /38112719xxxx/

rule 10 /210/ /38112719xxxx/

voice translation-profile SIP-OUTGOING

translate calling 1

class-map match-all L3-to-L2_VoIP-Cntrl

match ip dscp af31

class-map match-all L3-to-L2_VoIP-RTP

match ip dscp ef

class-map match-all SIP

match protocol sip

class-map match-all RTP

match protocol rtp

!

policy-map output-L3-to-L2

class L3-to-L2_VoIP-RTP

  set cos 5

class L3-to-L2_VoIP-Cntrl

  set cos 3

policy-map EthOut

class RTP

dial-peer cor list POZIV

member medjunarodni

sip-ua

credentials number 38112719xxxx username

38112719xxxx@ims.telek

ekomsrbija.com

registrar dns:ims.telekomsrbija.com expires 3600

sip-server ipv4:10.0.0.18:5060

host-registrar

refer-ood enable

handle-replaces

voice service voip
ip address trusted list
  ipv4 10.0.0.18 255.255.255.255
gcid
callmonitor
no cti shutdown
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fall
h323
  call start slow
modem passthrough nse codec g711alaw
sip
  header-passing
  registrar server expires max 1200 min 60
  early-offer forced
  midcall-signaling passthru
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw

voice translation-rule 1
rule 1 /201/ /xxxxx/
rule 2 /202/ /xxxx/
rule 4 /204/ /xxxxxx/
rule 5 /205/ /xxxxxx/
rule 6 /206/ /xxxxxx/
...

voice translation-profile SIP-OUTGOING
translate calling 1

class-map match-all L3-to-L2_VoIP-Cntrl
match ip dscp af31
class-map match-all L3-to-L2_VoIP-RTP
match ip dscp ef
class-map match-all SIP
match protocol sip
class-map match-all RTP
match protocol rtp
!
policy-map output-L3-to-L2
class L3-to-L2_VoIP-RTP
  set cos 5
class L3-to-L2_VoIP-Cntrl
  set cos 3
policy-map EthOut
class RTP

sip-ua
credentials number xxxxxxxxxx username USERNAME  mmmmmm
registrar dns:ims.nnnnnnnnn.com expires 3600
sip-server ipv4:10.0.0.18:5060
host-registrar
refer-ood enable
handle-replaces

my service provider says that does not forward calls to me because I reject them, and returns a '487 request canced'

[paolo bevilacqua]

Take "debug ccsip message" with "term mon".

BTe, you have many commands that are not needed..

Also, you would do good in makeing your DID numberssame as PSTN numbers.

[acazarkov]

resault of ddebug ccsip message:

Aug 22 10:01:31.381: //3572/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.x.x.110:5060;branch=z9hG4bKDD03C
From: <>204@ims.nnnnnnnnnnn.com>;tag=35466F8-BDC
To: <>204@ims.nnnnnnnnnn.com>;tag=1cba941c05e242a30adb173dca9e914
Call-ID: AC1D9F02-9B511E3-8006AECC-63706A48
Timestamp: 1377165691
CSeq: 312 REGISTER
Content-Length: 0
P-Charging-Vector: icid-value=1cba941c05e242a30adb173dca60b42

SIP/2.0 487 Request Cancelled

[acazarkov]

how to configure routet to accest call?

I think that is the problem ,router answer on incoming call with a request to call and that's why does not accept it.

[paolo bevilacqua]

You have posted only a fragment, not the complete trace as needed, and it is from a registation attempt, not a call.

It indicated that you negleted cto configure no-reg for the ephone-dn, so they try to register to ITSP unnecessarily.

As indicated abouve, you should configure your PSTN number a secondary to an ephone-dn, and register to ITSP.

[acazarkov]

Ruter#debug ccsip error

SIP Call error tracing is enabled

RuterPozarevac#

Aug 22 23:46:30.265: //-1/E70A39B082C5/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:

MF: Not a Forked SIP leg..

SIP: (6762) Attribute mid, level 1 instance 1 not found.

Aug 22 23:46:30.265: //6762/E70A39B082C5/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Aug 22 23:46:30.265: //6762/E70A39B082C5/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry

Aug 22 23:46:30.265: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count:

Unable to set CHANNEL_COUNT for callid 6762

Aug 22 23:46:30.265: //6762/E70A39B082C5/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.

Aug 22 23:46:30.281: //6763/E70A39B082C5/SIP/Error/ccsip_ipip_media_forking_read_from_TDContainer:

MF: Unable to read data from TD Container..

Aug 22 23:46:30.281: //6763/E70A39B082C5/SIP/Error/ccsip_ipip_media_forking_forked_leg_config:

MF: TD container cannot be read/container is NULL. Setting of forked call leg failed..

Aug 22 23:46:30.285: //6763/E70A39B082C5/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:

MF: Not a Forked SIP leg..

Aug 22 23:46:30.285: //6763/E70A39B082C5/SIP/Error/sip_iwf_sip_copy_channelInfo_to_sdp:

We are either escalating, orno stream found for this m-line index:1

Aug 22 23:46:30.285: //6763/E70A39B082C5/SIP/Error/sipSPI_ipip_set_history_info_header:

ccb->src_addr_str is NULL

SIP: (6763) Group (a= group line) attribute, level 65535 instance 1 not found.

SIP: (6763) Group (a= group line) attribute, level 65535 instance 1 not found.

Aug 22 23:46:30.297: //-1/xxxxxxxxxxxx/SIP/Error/get_content_length:

Could not get Content-length

Aug 22 23:46:30.325: //6764/E70A39B082C5/SIP/Error/ccsip_ipip_media_forking_read_from_TDContainer:

MF: Unable to read data from TD Container..

Aug 22 23:46:30.325: //6764/E70A39B082C5/SIP/Error/ccsip_ipip_media_forking_forked_leg_config:

MF: TD container cannot be read/container is NULL. Setting of forked call leg failed..

Aug 22 23:46:30.325: //6764/E70A39B082C5/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:

MF: Not a Forked SIP leg..

Aug 22 23:46:30.325: //6764/E70A39B082C5/SIP/Error/sip_iwf_sip_copy_channelInfo_to_sdp:

We are either escalating, orno stream found for this m-line index:1

SIP: (6764) Group (a= group line) attribute, level 65535 instance 1 not found.

SIP: (6764) Group (a= group line) attribute, level 65535 instance 1 not found.

Aug 22 23:46:30.333: //-1/xxxxxxxxxxxx/SIP/Error/get_content_length:

Could not get Content-length

Aug 22 23:46:40.185: //6765/000000000000/SIP/Error/ccsip_api_register_result_ind:

Message Code Class 4xx Method Code 100 received for REGISTER

Aug 22 23:46:40.185: //6765/000000000000/SIP/Error/sipSPIRegPthruProcessResponse:

Error NO RPCB

Aug 22 23:46:40.277: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_set_release_source_for_peer:

Failed AV set

Aug 22 23:46:41.013: //6766/000000000000/SIP/Error/ccsip_api_register_result_ind:

Message Code Class 4xx Method Code 100 received for REGISTER

Aug 22 23:46:41.013: //6766/000000000000/SIP/Error/sipSPIRegPthruProcessResponse:

Error NO RPCB

My conf on router :

voice-card 0

!

!

!

voice service voip

ip address trusted list

  ipv4 10.0.0.18 255.255.255.255

gcid

callmonitor

no cti shutdown

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

h323

  call start slow

modem passthrough nse codec g711alaw

sip

  header-passing

  registrar server expires max 1200 min 60

  early-offer forced

  midcall-signaling passthru

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

!

!

!

!

voice translation-rule 1

rule 1 /201/ /38112719xxxx/

...

!

!

voice translation-profile SIP-OUTGOING

translate calling 1

!

!

!

!

application

service toroute flash0:toroute.tcl

!

!

dial-peer cor custom

name mobilni

name fiksni

dial-peer cor list POZIVmobilni

member mobilni

!

dial-peer cor list POZIVfiksni

member fiksni

dial-peer cor list Mobil

member mobilni

member fiksni

dial-peer voice 1000 voip

corlist incoming Mobil

service toroute

destination-pattern [0-9]T

session protocol sipv2

session target sip-server

incoming called-number 38112719xxxx

voice-class codec 1

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

sip-ua

credentials number 38112719xxxx username 38112719xxxx@ims.nnnnnnn.com password 7 010107520F53515E75 realm ims.nnnnnnnn.com

registrar dns:ims.nnnnn.com expires 3600

sip-server ipv4:10.0.0.18:5060

host-registrar

refer-ood enable

handle-replaces

!

ephone-dn  1  dual-line

number 201

label 201

name office

corlist incoming Mobil

ephone  1

device-security-mode none

video

mac-address B4E9.B001.xxx

max-calls-per-button 2

username "user1" password 201

type 6921

button  1:1

[paolo bevilacqua]

Please post only the debug requested above, and nothing else.

[acazarkov]

Ruter#debug ccsip mess
SIP Call messages tracing is enabled
RuterPozarevac#
Aug 23 10:57:41.482: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:38112719xxxx@10.1.7.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.18:5060;branch=z9hG4bKk3p5jk100051nnc7g4o1.1
To: "firma"<>;cscf
From: <>0642947904@tssbg01.telekomsrbija.com;user=phone>;tag=279605674-1377248266952-
Call-ID: BW105746951230813315702207@10.10.1.143
CSeq: 355624293 INVITE
Max-Forwards: 8
Content-Length: 238
Contact: <064294XXXX>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY
Accept: multipart/mixed
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-hotsip-FileTransfer+xml
Supported: timer
P-Asserted-Identity: <64294XXXX>phone-context=+381@tssbg01.telekomsrbija.com;user=phone>
Privacy: none
P-Charging-Vector: icid-value=65685546060b32380adb6cda6173155
Min-SE: 180
Session-Expires: 1800
P-Called-Party-ID: <>

v=0
o=BroadWorks 80540950 1 IN IP4 10.0.0.18
s=-
c=IN IP4 10.0.0.18
t=0 0
a=sendrecv
m=audio 30964 RTP/AVP 8 18 98
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-15

Aug 23 10:57:41.494: //9292/AA8F88F98342/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.18:5060;branch=z9hG4bKk3p5jk100051nnc7g4o1.1
From: <>064294xxxx@tssbg01.telekomsrbija.com;user=phone>;tag=279605674-1377248266952-
To: "firma"<>;cscf
Date: Fri, 23 Aug 2013 10:57:41 GMT
Call-ID: BW105746951230813315702207@10.10.1.143
CSeq: 355624293 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Content-Length: 0

Aug 23 10:57:41.506: //9293/AA8F88F98342/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:100@10.0.0.18:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.110:5060;branch=z9hG4bK23FB1668
Remote-Party-ID: <064294XXXX>;party=calling;screen=no;privacy=off
From: sip:064294xxxx@tssbg01.telekomsrbija.com;tag=8AE2F98-146A
To: <100>
Date: Fri, 23 Aug 2013 10:57:41 GMT
Call-ID: AA91F999-B1911E3-8348AECC-63706A48@10.1.7.110
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2861533433-0186192355-2202185420-1668311624
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M3
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1377255461
Contact: <064294XXXX>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 7
Cisco-Gcid: AA8F88F9-0B19-11E3-8345-AECC63706A48
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 290

v=0
o=CiscoSystemsSIP-GW-UserAgent 6890 5773 IN IP4 10.1.7.110
s=SIP Call
c=IN IP4 10.1.7.110
t=0 0
m=audio 16644 RTP/AVP 8 100 101
c=IN IP4 10.1.7.110
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Aug 23 10:57:41.510: //9293/AA8F88F98342/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.7.110:5060;branch=z9hG4bK23FB1668
From: sip:064294xxxx@tssbg01.telekomsrbija.com;tag=8AE2F98-146A
To: <100>
Call-ID: AA91F999-B1911E3-8348AECC-63706A48@10.1.7.110
CSeq: 101 INVITE
Timestamp: 1377255461

Aug 23 10:57:41.538: //9293/AA8F88F98342/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.1.7.110:5060;branch=z9hG4bK23FB1668
From: sip:064294xxxx@tssbg01.telekomsrbija.com;tag=8AE2F98-146A
To: <100>;tag=832601292-1377248267013
Call-ID: AA91F999-B1911E3-8348AECC-63706A48@10.1.7.110
CSeq: 101 INVITE
Timestamp: 1377255461
Content-Length: 0
P-Charging-Vector: icid-value=f13ab6fa060987e10adb6cda65459f2

Aug 23 10:57:41.546: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:100@10.0.0.18:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.110:5060;branch=z9hG4bK23FB1668
From: sip:064294xxxx@tssbg01.telekomsrbija.com;tag=8AE2F98-146A
To: <100>;tag=832601292-1377248267013
Date: Fri, 23 Aug 2013 10:57:41 GMT
Call-ID: AA91F999-B1911E3-8348AECC-63706A48@10.1.7.110
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

Aug 23 10:57:41.550: //9294/AA8F88F98342/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:100@ims.telekomsrbija.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.110:5060;branch=z9hG4bK23FC9C1
Remote-Party-ID: <0642947904>;party=calling;screen=no;privacy=off
From: sip:064294xxxx@tssbg01.telekomsrbija.com;tag=8AE2FC4-A3C
To: <>100@ims.telekomsrbija.com>
Date: Fri, 23 Aug 2013 10:57:41 GMT
Call-ID: AA98145A-B1911E3-834AAECC-63706A48@79.101.99.70
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2861533433-0186192355-2202185420-1668311624
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M3
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1377255461
Contact: <064294XXXX>
Call-Info: <10.1.7.110:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 7
Cisco-Gcid: AA8F88F9-0B19-11E3-8345-AECC63706A48
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 314

v=0
o=CiscoSystemsSIP-GW-UserAgent 6362 7239 IN IP4 10.1.7.110
s=SIP Call
c=IN IP4 10.1.7.110
t=0 0
m=audio 16646 RTP/AVP 8 100 101 19
c=IN IP4 10.1.7.110
a=rtpmap:8 PCMA/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20

Aug 23 10:57:41.554: //9294/AA8F88F98342/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.7.110:5060;branch=z9hG4bK23FC9C1
From: sip:064294xxxx@tssbg01.telekomsrbija.com;tag=8AE2FC4-A3C
To: <>100@ims.telekomsrbija.com>
Call-ID: AA98145A-B1911E3-834AAECC-63706A48@79.101.99.70
CSeq: 101 INVITE
Timestamp: 1377255461

Aug 23 10:57:41.582: //9294/AA8F88F98342/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.1.7.110:5060;branch=z9hG4bK23FC9C1
From: sip:064294xxxx@tssbg01.telekomsrbija.com;tag=8AE2FC4-A3C
To: <>100@ims.telekomsrbija.com>;tag=1297836501-1377248267055
Call-ID: AA98145A-B1911E3-834AAECC-63706A48@79.101.99.70
CSeq: 101 INVITE
Timestamp: 1377255461
Content-Length: 0
P-Charging-Vector: icid-value=f8615e8b05ee31d10adb6cda67a645b

Aug 23 10:57:41.586: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:100@ims.telekomsrbija.com:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.110:5060;branch=z9hG4bK23FC9C1
From: sip:064294xxxx@tssbg01.telekomsrbija.com;tag=8AE2FC4-A3C
To: <>100@ims.telekomsrbija.com>;tag=1297836501-1377248267055
Date: Fri, 23 Aug 2013 10:57:41 GMT
Call-ID: AA98145A-B1911E3-834AAECC-63706A48@79.101.99.70
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

Aug 23 10:57:51.502: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:38112719xxxx@10.1.7.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.18:5060;branch=z9hG4bKk3p5jk100051nnc7g4o1.1
CSeq: 355624293 CANCEL
To: "firma"<>;cscf
From: <>064294xxxx@tssbg01.telekomsrbija.com;user=phone>;tag=279605674-1377248266952-
Call-ID: BW105746951230813315702207@10.10.1.143
Max-Forwards: 8
Content-Length: 0

Aug 23 10:57:51.502: //9292/AA8F88F98342/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.18:5060;branch=z9hG4bKk3p5jk100051nnc7g4o1.1
From: <>064294xxxx@tssbg01.telekomsrbija.com;user=phone>;tag=279605674-1377248266952-
To: "firma"<>;cscf
Date: Fri, 23 Aug 2013 10:57:51 GMT
Call-ID: BW105746951230813315702207@10.10.1.143
CSeq: 355624293 CANCEL
Content-Length: 0

Aug 23 10:57:51.506: //9292/AA8F88F98342/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.0.18:5060;branch=z9hG4bKk3p5jk100051nnc7g4o1.1
From: <>064294xxxx@tssbg01.telekomsrbija.com;user=phone>;tag=279605674-1377248266952-
To: "firma"<>;cscf;tag=8AE56A8-2407
Date: Fri, 23 Aug 2013 10:57:51 GMT
Call-ID: BW105746951230813315702207@10.10.1.143
CSeq: 355624293 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Reason: Q.850;cause=16
Content-Length: 0

Aug 23 10:57:51.510: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:381127195224@10.1.7.110:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.18:5060;branch=z9hG4bKk3p5jk100051nnc7g4o1.1
CSeq: 355624293 ACK
To: "firma"<>;cscf;tag=8AE56A8-2407
From: <>064294xxxx@tssbg01.telekomsrbija.com;user=phone>;tag=279605674-1377248266952-
Call-ID: BW105746951230813315702207@10.10.1.143
Max-Forwards: 8
Content-Length: 0

[acazarkov]

Ruter#debug ccsip call
Aug 23 11:37:48.010: //9457/44ED61D0835E/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x23809A88
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 064294xxxx
Called Number            : 100
Source IP Address (Sig  ): 10.1.7.110
Destn SIP Req Addr:Port  : 10.0.0.18:5060
Destn SIP Resp Addr:Port : 10.0.0.18:5060
Destination Name         : ims.telekomsrbija.com

Aug 23 11:37:48.010: //9457/44ED61D0835E/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): 10.1.7.110
Source IP Port    (Media): 16656
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

Aug 23 11:37:48.010: //9457/44ED61D0835E/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 57
Disconnect Cause (SIP)   : 403

Aug 23 11:37:48.054: //9458/44ED61D0835E/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x238218E8
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 064294xxxx
Called Number            : 100
Source IP Address (Sig  ): 10.1.7.110
Destn SIP Req Addr:Port  : 10.0.0.18:5060
Destn SIP Resp Addr:Port : 10.0.0.18:5060
Destination Name         : 10.0.0.18

Aug 23 11:37:48.054: //9458/44ED61D0835E/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): 10.1.7.110
Source IP Port    (Media): 16658
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

Aug 23 11:37:48.054: //9458/44ED61D0835E/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 57
Disconnect Cause (SIP)   : 403

Aug 23 11:37:57.990: //9456/44ED61D0835E/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x2380FA20
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 064294xxx
Called Number            : 38112719xxxx
Source IP Address (Sig  ): 10.1.7.110
Destn SIP Req Addr:Port  : 10.0.0.18:5060
Destn SIP Resp Addr:Port : 10.0.0.18:5060
Destination Name         : 10.0.0.18

Aug 23 11:37:57.990: //9456/44ED61D0835E/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711alaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 8 (tx), 8 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 98 (tx), 98 (rx)
Source IP Address (Media): 10.1.7.110
Source IP Port    (Media): 16654
Destn  IP Address (Media): 10.0.0.18
Destn  IP Port    (Media): 34918
Orig Destn IP Address:Port (Media): [ - ]:0

Aug 23 11:37:57.990: //9456/44ED61D0835E/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 487