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New Member

Problem with external na call- CISCO 2801 -pbx

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?

2 ACCEPTED SOLUTIONS

Accepted Solutions
Hall of Fame Super Gold

Re: Problem with external na call- CISCO 2801 -pbx

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.

Problem with external na call- CISCO 2801 -pbx

Hi,

You created outgoing translation rule to replace the calling number from 20x to 38112719xxxx

Now, you create another translation rule for incoming calls which replace the called 38112719xxxx to 20x as below

voice translation-rule 2

rule 1 /38112719xxxx/ /201/

rule 2 /38112719xxxx/ /202/

voice translation-profile SIP-INCOMING

translate called 2

dial-peer voice xxx voip

translation-profile incoming SIP-INCOMING

Note: the secondary number is not needed here, the call will land to the respective inetrnal number.

Regards

Selvarathnam

31 REPLIES
Hall of Fame Super Gold

Problem with external na call- CISCO 2801 -pbx

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

In simple words, what that means?

New Member

Problem with external na call- CISCO 2801 -pbx

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

Hall of Fame Super Gold

Problem with external na call- CISCO 2801 -pbx

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

New Member

Problem with external na call- CISCO 2801 -pbx

It is SIP TRUNK.

Hall of Fame Super Gold

Re: Problem with external na call- CISCO 2801 -pbx

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.

New Member

Re: Problem with external na call- CISCO 2801 -pbx

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'

Hall of Fame Super Gold

Re: Problem with external na call- CISCO 2801 -pbx

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.

New Member

Re: Problem with external na call- CISCO 2801 -pbx

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

New Member

Re: Problem with external na call- CISCO 2801 -pbx

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.

Hall of Fame Super Gold

Re: Problem with external na call- CISCO 2801 -pbx

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.

New Member

Re: Problem with external na call- CISCO 2801 -pbx

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


Hall of Fame Super Gold

Re: Problem with external na call- CISCO 2801 -pbx

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

New Member

Re: Problem with external na call- CISCO 2801 -pbx

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

New Member

Re: Problem with external na call- CISCO 2801 -pbx

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

Hall of Fame Super Gold

Re: Problem with external na call- CISCO 2801 -pbx

As indicated above already, you need to configure 38112719xxxx as secondary number to the phone where you want the call to ring.

New Member

Re: Problem with external na call- CISCO 2801 -pbx

ok, can you help me with that?

Hall of Fame Super Gold

Problem with external na call- CISCO 2801 -pbx

Just do what is indicated above.

New Member

Re: Problem with external na call- CISCO 2801 -pbx

how to configure secondary number to the phone?

Hall of Fame Super Gold

Re: Problem with external na call- CISCO 2801 -pbx

Using 'secondary' in 'number' command.

You can reference the mnual for more information.

Problem with external na call- CISCO 2801 -pbx

Hi,

You created outgoing translation rule to replace the calling number from 20x to 38112719xxxx

Now, you create another translation rule for incoming calls which replace the called 38112719xxxx to 20x as below

voice translation-rule 2

rule 1 /38112719xxxx/ /201/

rule 2 /38112719xxxx/ /202/

voice translation-profile SIP-INCOMING

translate called 2

dial-peer voice xxx voip

translation-profile incoming SIP-INCOMING

Note: the secondary number is not needed here, the call will land to the respective inetrnal number.

Regards

Selvarathnam

Hall of Fame Super Gold

Re: Problem with external na call- CISCO 2801 -pbx

Note: the secondary number is not needed here, the call will land to the respective inetrnal number.

The point is: secondary number configuration allow to avoid all translations with a single, simple command.

Normally, translation is used when there is a range of DID, but often SIP trunk subscribers only have few discontinuous numbers, so the secondary approach if preferable.

Then each one can choose what he wants.

New Member

Re: Problem with external na call- CISCO 2801 -pbx

I remove my old dial-peer voice 5000 voip an add:

voice translation-rule 2

rule 1 /38112719xxxx/ /201/

rule 2 /38112719xxxx/ /202/

voice translation-profile SIP-INCOMING

translate called 2

dial-peer voice 999 voip

translation-profile incoming SIP-INCOMING

corlist incoming Mobil

session protocol sipv2

session target sip-server

incoming called-number 38112719xxxx

I still have a problem:

Ruter#debug ccsip error

SIP Call error tracing is enabled

Ruter#

Aug 23 19:07:14.288: //-1/0E1D6E6283AC/SIP/Error/ccsip_ipip_media_forking_update_preferred_codec:

MF: Not a Forked SIP leg..

SIP: (11163) Attribute mid, level 1 instance 1 not found.

Aug 23 19:07:14.288: //11163/0E1D6E6283AC/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:

failed to update call entry

Aug 23 19:07:14.288: //11163/0E1D6E6283AC/SIP/Error/sipSPI_ipip_update_call_entry:

failed to update call entry

Aug 23 19:07:14.288: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count:

Unable to set CHANNEL_COUNT for callid 11163

Aug 23 19:07:14.288: //11163/0E1D6E6283AC/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 23 19:07:14.300: //11163/0E1D6E6283AC/SIP/Error/sipSPI_ipip_set_history_info_header:

Not SIP2SIP mode

Aug 23 19:07:29.304: //11163/0E1D6E6283AC/SIP/Error/sipSPI_ipip_set_history_info_header:

Not SIP2SIP mode

Aug 23 19:07:29.308: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_set_release_source_for_peer:

Failed AV set

Aug 23 19:07:29.308: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG:

No Inbound Container Created !!!

Aug 23 19:07:29.308: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931:

No Inbound Container Created !!!

Aug 23 19:07:52.532: //11165/000000000000/SIP/Error/ccsip_api_register_result_ind:

Message Code Class 4xx Method Code 100 received for REGISTER

Aug 23 19:07:52.532: //11165/000000000000/SIP/Error/sipSPIRegPthruProcessResponse:

Error NO RPCB

Aug 23 19:07:52.532: //11166/000000000000/SIP/Error/ccsip_api_register_result_ind:

Message Code Class 4xx Method Code 100 received for REGISTER

Aug 23 19:07:52.532: //11166/000000000000/SIP/Error/sipSPIRegPthruProcessResponse:

Error NO RPCB

Aug 23 19:07:52.536: //11167/000000000000/SIP/Error/ccsip_api_register_result_ind:

Message Code Class 4xx Method Code 100 received for REGISTER

Aug 23 19:07:52.536: //11167/000000000000/SIP/Error/sipSPIRegPthruProcessResponse:

Error NO RPCB

Aug 23 19:07:52.568: //11168/000000000000/SIP/Error/ccsip_api_register_result_ind:

Message Code Class 4xx Method Code 100 received for REGISTER

Aug 23 19:07:52.568: //11168/000000000000/SIP/Error/sipSPIRegPthruProcessResponse:

Error NO RPCB

Aug 23 19:07:52.624: //11169/000000000000/SIP/Error/ccsip_api_register_result_ind:

Message Code Class 4xx Method Code 100 received for REGISTER

Aug 23 19:07:52.624: //11169/000000000000/SIP/Error/sipSPIRegPthruProcessResponse:

Error NO RPCB

Aug 23 19:07:52.804: //11170/000000000000/SIP/Error/ccsip_api_register_result_ind:

Message Code Class 4xx Method Code 100 received for REGISTER

Aug 23 19:07:52.804: //11170/000000000000/SIP/Error/sipSPIRegPthruProcessResponse:

Error NO RPCB

New Member

Re: Problem with external na call- CISCO 2801 -pbx

Ruter#DEBUG CCSIP CALL

SIP Call statistics tracing is enabled

Ruter#

Aug 23 19:26:13.128: //11244/ABF6220D83B3/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            : 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 19:26:13.128: //11244/ABF6220D83B3/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g729r8

Negotiated Codec Bytes   : 20

Nego. Codec payload      : 18 (tx), 18 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 10.1.7.110

Source IP Port    (Media): 16686

Destn  IP Address (Media): 10.0.0.18

Destn  IP Port    (Media): 32562

Orig Destn IP Address:Port (Media): [ - ]:0

Aug 23 19:26:13.128: //11244/ABF6220D83B3/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 16

Disconnect Cause (SIP)   : 302

New Member

Re: Problem with external na call- CISCO 2801 -pbx

and I added a command

  voice service voip

   no supplementary-service sip moved-temporarily

and finally I established a connection.

I am so happy

New Member

Re: Problem with external na call- CISCO 2801 -pbx

Now I have another problem. When I call number 38112719xxxb it rings on both phones 38112719xxxa (201) and  38112719xxxb (202)


voice translation-rule 2

rule 1 /38112719xxxa/ /201/

rule 2 /38112719xxxb/ /202/

voice translation-profile SIP-INCOMING

translate called 2

dial-peer cor custom

name mobilni

dial-peer cor list Mobil

member mobilni

dial-peer voice 999 voip

corlist incoming Mobil

translation-profile incoming SIP-INCOMING

session protocol sipv2

session target sip-server

incoming called-number 381127195xxxa

dial-peer voice 991 voip

corlist incoming Mobil

translation-profile incoming SIP-INCOMING

session protocol sipv2

session target sip-server

incoming called-number 38112719xxxb

ephone-dn  1  dual-line

number 201

label 201

description  201

name Info 201

corlist incoming Mobil

!

!

ephone-dn  2  dual-line

number 202

label 202

description  202

name Office 202

corlist incoming Mobil

ephone  1

device-security-mode none

video

mac-address B4E9.B001.cccc

max-calls-per-button 2

username "user1" password 201

type 6921

button  1:1

!

!

!

ephone  2

device-security-mode none

video

mac-address B4E9.B001.yyyy

max-calls-per-button 2

username "user2" password 202

type 6921

button  1:2

Re: Problem with external na call- CISCO 2801 -pbx

Why you required two dial-peer 999 and 991

no need of incoming called-number command or change it to incoming called-number 38112719....

see how many dial-peer it is created:

#show telephony-service dial-peer

New Member

Re: Problem with external na call- CISCO 2801 -pbx

I remove dial-peer 991 and change incoming number to 38112719....

but still the same problem

Rute#sh telephony-service dial-peer

dial-peer voice 20001 pots

destination-pattern 201$

huntstop

corlist incoming Mobil

progress_ind setup enable 3

port 50/0/1

dial-peer voice 20002 pots

destination-pattern 202$

huntstop

corlist incoming Mobil

progress_ind setup enable 3

port 50/0/2

Re: Problem with external na call- CISCO 2801 -pbx

Please, provide us with the full running-configuration of the router.

New Member

Re: Problem with external na call- CISCO 2801 -pbx

ip dhcp pool VOICE

network 10.10.108.0 255.255.255.0

default-router 10.10.108.1

option 150 ip 10.10.108.1

!

!

!

ip domain name ims.nnnnnnnnn.com

ip host ims.nnnnnnnn.com 10.0.0.18

no ipv6 cef

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

no supplementary-service sip moved-temporarily

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/

rule 2 /202/ /38112719xxx1/

rule 3 /203/ /38112719xxx2/

rule 4 /204/ /38112719xxx3/

rule 5 /205/ /38112719xxx4/

rule 6 /206/ /38112719xxx5/

rule 7 /207/ /38112719xxx6/

rule 8 /208/ /38112719xxx7/

rule 9 /209/ /38112719xxx8/

rule 10 /210/ /38112719xxx9/

rule 11 /211/ /38112719xx10/

!

voice translation-rule 2

rule 1 /38112719xxxx/ /201/

rule 2 /38112719xxx1/ /202/

rule 3 /38112719xxx2/ /201/

rule 4 /38112719xxx3/ /204/

rule 5 /38112719xxx4/ /205/

rule 6 /38112719xxx5/ /206/

rule 7 /38112719xxx6/ /207/

rule 8 /38112719xxx7/ /208/

rule 9 /38112719xxx8/ /209/

rule 10 /38112719xxx9/ /210/

rule 11 /38112719xx10/ /211/

!

!

voice translation-profile SIP-INCOMING

translate called 2

!

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 custom

name medjunarodni

name mobilni

name fiksni

name lokalni

name besplatni

name igrenasrecu

name sanaplatom

name hitnesluzbe

name servisnesluzbe

!

!

dial-peer cor list POZIVmedjunarodni

member medjunarodni

!

dial-peer cor list POZIVmobilni

member mobilni

!

dial-peer cor list POZIVfiksni

member fiksni

!

dial-peer cor list POZIVlokalni

member lokalni

!

dial-peer cor list POZIVbesplatni

member besplatni

!

dial-peer cor list POZIVsanaplatom

member sanaplatom

!

dial-peer cor list POZIVigrenasrecu

member igrenasrecu

!

dial-peer cor list POZIVhitnesluzbe

member hitnesluzbe

!

dial-peer cor list POZIVservisnesluzbe

member servisnesluzbe

!

dial-peer cor list SviPozivi

member medjunarodni

member mobilni

member fiksni

member lokalni

member besplatni

member igrenasrecu

member sanaplatom

member hitnesluzbe

member servisnesluzbe

!

dial-peer cor list Mobil

member mobilni

member fiksni

member lokalni

member besplatni

member hitnesluzbe

member servisnesluzbe

!

dial-peer cor list Fiksni

member fiksni

member lokalni

member besplatni

member hitnesluzbe

!

!

dial-peer voice 1001 voip

corlist outgoing POZIVmedjunarodni

destination-pattern 00T

session protocol sipv2

session target dns:ims.nnnnnnn.com

dtmf-relay sip-notify rtp-nte

codec g711alaw

!

dial-peer voice 1002 voip

corlist outgoing POZIVfiksni

translation-profile outgoing SIP-OUTGOING

destination-pattern 0[1-3]T

modem passthrough nse codec g711alaw

session protocol sipv2

session target dns:ims.nnnnnnn.com

dtmf-relay sip-notify rtp-nte

codec g711alaw

fax rate disable

no vad

!

dial-peer voice 1003 voip

corlist outgoing POZIVigrenasrecu

translation-profile outgoing SIP-OUTGOING

destination-pattern 0[4-5]T

session protocol sipv2

session target dns:ims.nnnnn.com

dtmf-relay sip-notify rtp-nte

codec g711alaw

!

dial-peer voice 1004 voip

corlist outgoing POZIVmobilni

translation-profile outgoing SIP-OUTGOING

destination-pattern 06T

session protocol sipv2

session target dns:imsn.nnnnnn.com

dtmf-relay sip-notify rtp-nte

codec g711alaw

!

dial-peer voice 1005 voip

corlist outgoing POZIVsanaplatom

translation-profile outgoing SIP-OUTGOING

destination-pattern 0[79]T

session protocol sipv2

session target dns:ims.nnnnnnnnn.com

dtmf-relay sip-notify rtp-nte

codec g711alaw

!

dial-peer voice 1006 voip

corlist outgoing POZIVbesplatni

translation-profile outgoing SIP-OUTGOING

destination-pattern 0[8]T

session protocol sipv2

session target dns:ims.nnnnnnnn.com

dtmf-relay sip-notify rtp-nte

codec g711alaw

!

dial-peer voice 1007 voip

corlist outgoing POZIVlokalni

translation-profile outgoing SIP-OUTGOING

destination-pattern [1-8]T

session protocol sipv2

session target dns:ims.tnnnnnnn.com

dtmf-relay sip-notify rtp-nte

codec g711alaw

!

dial-peer voice 1008 voip

corlist outgoing POZIVhitnesluzbe

translation-profile outgoing SIP-OUTGOING

destination-pattern 9[2-4]T

session protocol sipv2

session target dns:ims.nnnnnnn.com

dtmf-relay sip-notify rtp-nte

codec g711alaw

!

dial-peer voice 1009 voip

corlist outgoing POZIVservisnesluzbe

translation-profile outgoing SIP-OUTGOING

destination-pattern 9[5-8]T

session protocol sipv2

session target dns:ims.nnnnnn.com

dtmf-relay sip-notify rtp-nte

codec g711alaw

!

dial-peer voice 999 voip

corlist incoming Mobil

translation-profile incoming SIP-INCOMING

session protocol sipv2

session target sip-server

incoming called-number 38112719....

!

!

sip-ua

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

registrar dns:ims.nnnnn.com expires 3600

sip-server ipv4:10.0.0.18:5060

host-registrar

refer-ood enable

handle-replaces

!

!

!

gatekeeper

no shutdown

!

!

telephony-service

no auto-reg-ephone

pin 11319 override

max-ephones 24

max-dn 72

ip source-address 10.10.108.1 port 2000

calling-number initiator

service phone videoCapability 1

timeouts interdigit 5

system message MESSAGE

cnf-file location flash:

time-zone 28

time-format 24

date-format dd-mm-yy

max-conferences 8 gain -6

call-forward pattern .T

moh "music-on-hold.au"

dn-webedit

time-webedit

transfer-system full-consult dss

create cnf-files version-stamp 7960 Aug 21 2013 22:05:22

!

!

ephone-dn  1  dual-line

number 201

label 201

description  201

name 201

corlist incoming Mobil

!

!

ephone-dn  2  dual-line

number 202

label 202

description OFFICE 202

name Office 202

corlist incoming Mobil

!

!

ephone-dn  3  dual-line

number 203

label 203

corlist incoming Mobil

!

!

ephone-dn  4  dual-line

number 204

label 204

corlist incoming Mobil

!

!

ephone-dn  5  dual-line

number 205

label 205

description  205

name 205

corlist incoming Mobil

!

!

ephone-dn  6  dual-line

number 206

label 206

description 206

corlist incoming Mobil

!

!

ephone-dn  7  dual-line

number 207

label 207

name 207

corlist incoming Mobil

!

!

ephone-dn  8  dual-line

number 208

label 208

description 208

name 208

corlist incoming Mobil

!

!

ephone-dn  9  dual-line

number 209

label 209

description 209

name 209

corlist incoming Mobil

!

!

ephone-dn  10  dual-line

number 210

label 210

description 210

name 210

corlist incoming Mobil

!

!

ephone-dn  11  dual-line

number 211

label 211

description 211

name 211

corlist incoming Mobil

!

ephone  1

device-security-mode none

video

mac-address B4E9.B001.xxxx

max-calls-per-button 2

username "user1" password 201

type 6921

button  1:1

!

!

!

ephone  2

device-security-mode none

video

mac-address B4E9.B001.vvvv

max-calls-per-button 2

username "user2" password 202

type 6921

button  1:2

!

!

!

ephone  3

device-security-mode none

video

username "user3" password 203

!

!

!

ephone  4

device-security-mode none

video

mac-address B4E9.B001.gggg

max-calls-per-button 2

username "user4" password 204

type 6921

button  1:4

!

!

!

ephone  5

device-security-mode none

video

mac-address B4E9.B001.hhhh

max-calls-per-button 2

username "user5" password 205

type 6921

button  1:5

!

!

!

ephone  7

device-security-mode none

video

mac-address B4E9.B001.jjjjj

max-calls-per-button 2

username "user7" password 207

type 6921

button  1:7

!

!

!

ephone  8

device-security-mode none

video

mac-address B4E9.B001.hhhh

max-calls-per-button 2

username "user8" password 208

type 6921

button  1:8

!

!

!

ephone  9

device-security-mode none

video

mac-address B4E9.B001.jiii

max-calls-per-button 2

username "user9" password 209

type 6921

button  1:9

!

!

!

ephone  10

device-security-mode none

video

mac-address B4E9.B001.rrrr

max-calls-per-button 2

username "user10" password 210

type 6921

button  1:10

!

!

!

ephone  11

device-security-mode none

video

max-calls-per-button 2

username "user11" password 211

type 6921

!

!

!

!

line con 0

login local

line aux 0

line 2

--More--

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