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New Member

forward all not working

hi all!

When i activate Forward ALL on Cisco Phone to number that begins 8... forward works fine . IF i use 08.. (08 calls going through another department )  i listen silence.

Calls through 08 without Forward ALL working fine 

I  write rules  to Forward all internal calls to my mobile phone but its not helps me . 

voice translation-rule 200
rule 1 /\(^68..\)/ /08_my_mobile_phone_number\1/   after that rule try rule 1 /^68../ /08_my_mobile_phone_number/

voice translation-profile 1
translate calling 200

ephone-dn 1

translation-profile outgoing 1

sh call history voice brief shows:

879  : 1980520 75536864ms.5231 (10:50:53.597 Russia Wed Jul 30 2014) +-1 +980 pid:40003 Answer 6802
 dur 00:00:00 tx:0/0 rx:0/0 10  (normal call clearing (16)) dscp:0 media:0 audio tos:0x0 video tos:0x0
 IP IP_PHONE:20686 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded No
  media inactive detected:n media contrl rcvd:n/a timestamp:n/a
  long duration call detected:n long dur callduration :n/a timestamp:n/a

 LostPacketRate:0.00 OutOfOrderRate:0.00
87F  : 1980523 75537864ms.5232 (10:50:54.597 Russia Wed Jul 30 2014) +-1 +50 pid:1002 Originate 08_my_mobile_phone_number
 dur 00:00:00 tx:0/0 rx:0/0 1   (unassigned number (1)) dscp:0 media:0 audio tos:0xB8 video tos:0x0
 IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded No
  media inactive detected:n media contrl rcvd:n/a timestamp:n/a
  long duration call detected:n long dur callduration :n/a timestamp:n/a

 

 

Everyone's tags (1)
1 ACCEPTED SOLUTION

Accepted Solutions

When sending the Invite out

When sending the Invite out to the other department, you're getting a 404 Not Found from them.

First attempt:

018708: Jul 30 18:11:11.245: //1994314/315DCB75B7F5/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:mobile_phone@172.31.5.2:5060 SIP/2.0
Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3C523CC17
Remote-Party-ID: "NAME" <sip:6820@CISCO_IP>;party=calling;screen=yes;privacy=off
From: "NAME" <sip:6820@CISCO_IP>;tag=613B3E8-2028
To: <sip:mobile_phone@172.31.5.2>
Date: Wed, 30 Jul 2014 14:11:11 GMT
Call-ID: 31603CAD-172A11E4-B7FBEC04-71085274@CISCO_IP
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0828230517-0388633060-3086347268-1896370804
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1406729471
Contact: <sip:6820@CISCO_IP:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Diversion: <sip:6811@CISCO_IP>;reason=unconditional;counter=1
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4795 2908 IN IP4 CISCO_IP
s=SIP Call
c=IN IP4 CISCO_IP
t=0 0
m=audio 25022 RTP/AVP 0 101 19
c=IN IP4 CISCO_IP
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

It sends the Invite again:

018711: Jul 30 18:11:11.741: //1994314/315DCB75B7F5/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:mobile_phone@172.31.5.2:5060 SIP/2.0
Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3C523CC17
Remote-Party-ID: "NAME" <sip:6820@CISCO_IP>;party=calling;screen=yes;privacy=off
From: "NAME" <sip:6820@CISCO_IP>;tag=613B3E8-2028
To: <sip:mobile_phone@172.31.5.2>
Date: Wed, 30 Jul 2014 14:11:11 GMT
Call-ID: 31603CAD-172A11E4-B7FBEC04-71085274@CISCO_IP
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0828230517-0388633060-3086347268-1896370804
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1406729471
Contact: <sip:6820@CISCO_IP:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Diversion: <sip:6811@CISCO_IP>;reason=unconditional;counter=1
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4795 2908 IN IP4 CISCO_IP
s=SIP Call
c=IN IP4 CISCO_IP
t=0 0
m=audio 25022 RTP/AVP 0 101 19
c=IN IP4 CISCO_IP
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

 

CME receives 100 Trying:

018712: Jul 30 18:11:11.761: //1994314/315DCB75B7F5/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Date: Wed, 30 Jul 2014 14:11:21 GMT
From: "NAME" <sip:6820@CISCO_IP>;tag=613B3E8-2028
Allow-Events: presence
Content-Length: 0
To: <sip:mobile_phone@172.31.5.2>
Call-ID: 31603CAD-172A11E4-B7FBEC04-71085274@CISCO_IP
Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3C523CC17
CSeq: 101 INVITE

 

CME sends 404 Not Found:

018713: Jul 30 18:11:11.785: //1994314/315DCB75B7F5/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 404 Not Found
Reason: Q.850;cause=1
Date: Wed, 30 Jul 2014 14:11:21 GMT
From: "NAME" <sip:6820@CISCO_IP>;tag=613B3E8-2028
Allow-Events: presence
Content-Length: 0
To: <sip:mobile_phone@172.31.5.2>;tag=ad803d66-05af-4540-8ebe-481e90667f53-20224504
Call-ID: 31603CAD-172A11E4-B7FBEC04-71085274@CISCO_IP
Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3C523CC17
CSeq: 101 INVITE

 

It may be because of the Diversion header in the Invite that the other side doesn't like or it may be because the number format is not correct.  You may want to compare to an Invite from a direct call that does work.

You could try removing the diversion header:

voice class sip-profiles 100
  request INVITE sip-header Diversion remove
voice service voip 
   sip
     sip-profiles 100

 

17 REPLIES
VIP Super Bronze

Can you share your sh run

Can you share your sh run here..Also can you attach the output of the following debug while testing the cfwd issue...

debug voip ccapi inout

debug isdn q931 (if you are using ISDN)

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

Thanks for your answer.I

Thanks for your answer.

I execute

debug voip ccapi inout    
voip ccapi inout debugging is on

After that i try testing call but no have any messages.

Second command not worked

debug isdn q931
                       ^
% Invalid input detected at '^' marker.

i put running conf but wait few minutes pls 

 

 

New Member

output of sh run  in

output of sh run  in attachment

VIP Super Bronze

Can you do anothere test and

Can you do anothere test and send the ff:

debug ccsip messages

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

i send term on and debug

i send term on and debug ccsip messages

 

 

6820 - from called
6811 - phone where Call Forwarding Enable to mobile_phone

Sent:
INVITE sip:mobile_phone@IP_of_other_department:5060 SIP/2.0
Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3C47391A8B
Remote-Party-ID: "NAME" <sip:6820@CISCO_IP>;party=calling;screen=yes;privacy=off
From: "NAME" <sip:6820@CISCO_IP>;tag=5E9F070-265B
To: <sip:CISCO_IP@IP_of_other_department>
Date: Wed, 30 Jul 2014 13:25:38 GMT
Call-ID: D1FD10FF-172311E4-B7E3EC04-71085274@CISCO_IP
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3509902782-0388174308-3084643332-1896370804
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1406726738
Contact: <sip:6820@CISCO_IP:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Diversion: <sip:6811@CISCO_IP>;reason=unconditional;counter=1
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 7250 5685 IN IP4 CISCO_IP
s=SIP Call
c=IN IP4 CISCO_IP
t=0 0
m=audio 25016 RTP/AVP 0 101 19
c=IN IP4 CISCO_IP
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

 

 


Sent:
INVITE sip:mobile_phone@IP_of_other_department SIP/2.0
Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3C47391A8B
Remote-Party-ID: "NAME" <sip:6820@CISCO_IP>;party=calling;screen=yes;privacy=off
From: "NAME" <sip:6820@CISCO_IP>;tag=5E9F070-265B
To: <sip:mobile_phone@IP_of_other_department>
Date: Wed, 30 Jul 2014 13:25:42 GMT
Call-ID: D1FD10FF-172311E4-B7E3EC04-71085274@172.23.1.2
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 3509902782-0388174308-3084643332-1896370804
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1406726742
Contact: <sip:6820@CISCO_IP:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Diversion: <sip:6811@CISCO_IP>;reason=unconditional;counter=1
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 7250 5685 IN IP4 CISCO_IP
s=SIP Call
c=IN IP4 CISCO_IP
t=0 0
m=audio 25016 RTP/AVP 0 101 19
c=IN IP4 CISCO_IP
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

Jul 30 13:25:42.675: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:mobile_phone@CISCO_IP:5060 SIP/2.0
Via: SIP/2.0/UDP PHONE_IP:5060;branch=z9hG4bK3d64955a
From: "NAME" <sip:6820@CISCO_IP>;tag=7c95f322ab0b0fd64d5cf396-06f405a6
To: <sip:6@CISCO_IP> ###### only 6
Call-ID: 7c95f322-ab0b00a1-5abe5a7f-7ef1f373@172.23.1.40
Max-Forwards: 70
Date: Wed, 30 Jul 2014 13:25:41 GMT
CSeq: 102 CANCEL
User-Agent: Cisco-CP8961/9.3.2
Content-Length: 0


Jul 30 13:25:42.675: //1992896/D134DDBEB7DB/CCAPI/cc_api_call_disconnected:
   Cause Value=16, Interface=0x3DC80904, Call Id=1992896
Jul 30 13:25:42.675: //1992896/D134DDBEB7DB/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=FALSE, Cause Value=16, Retry Count=0)
Jul 30 13:25:42.675: //1992898/D134DDBEB7DB/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Jul 30 13:25:42.675: //1992898/D134DDBEB7DB/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=FALSE, Cause Value=16)
Jul 30 13:25:42.675: //1992896/D134DDBEB7DB/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)
Jul 30 13:25:42.675: //1992896/D134DDBEB7DB/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Jul 30 13:25:42.679: //1992896/D134DDBEB7DB/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.1.40:5060;branch=z9hG4bK3d64955a
From: "NAME" <sip:6820@CISCO_IP>;tag=7c95f322ab0b0fd64d5cf396-06f405a6
To: <sip:6@CISCO_IP>
Date: Wed, 30 Jul 2014 13:25:42 GMT
Call-ID: 7c95f322-ab0b00a1-5abe5a7f-7ef1f373@PHONE_IP
CSeq: 102 CANCEL
Content-Length: 0


Jul 30 13:25:42.679: //1992896/D134DDBEB7DB/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP PHONE_IP:5060;branch=z9hG4bK3d64955a
From: "NAME" <sip:6820@CISCO_IP>;tag=7c95f322ab0b0fd64d5cf396-06f405a6
To: <sip:6@CISCO_IP>;tag=5EA1170-1428
Date: Wed, 30 Jul 2014 13:25:42 GMT
Call-ID: 7c95f322-ab0b00a1-5abe5a7f-7ef1f373@172.23.1.40
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.3.3.M
Reason: Q.850;cause=16
Content-Length: 0


Jul 30 13:25:42.759: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:mobile_phone@CISCO_IP:5060 SIP/2.0
Via: SIP/2.0/UDP PHONE_IP:5060;branch=z9hG4bK3d64955a
From: "NAME" <sip:6820@CISCO_IP>;tag=7c95f322ab0b0fd64d5cf396-06f405a6
To: <sip:6@CISCO_IP>;tag=5EA1170-1428
Call-ID: 7c95f322-ab0b00a1-5abe5a7f-7ef1f373@PHONE_IP
Max-Forwards: 70
Date: Wed, 30 Jul 2014 13:25:41 GMT
CSeq: 102 ACK
Content-Length: 0


Jul 30 13:25:42.759: //1992896/D134DDBEB7DB/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x3DC80904, Tag=0x0, Call Id=1992896,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Jul 30 13:25:42.763: //1992896/D134DDBEB7DB/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Jul 30 13:25:42.763: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
   
Jul 30 13:25:42.763: :cc_free_feature_vsa freeing 21013B50
Jul 30 13:25:42.763: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
   
Jul 30 13:25:42.763:  vsacount in free is 3
Jul 30 13:25:42.763: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
   
Jul 30 13:25:42.763:  vsacount in free is 2
Jul 30 13:25:42.763: //1992896/D134DDBEB7DB/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3CF18AD8
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 6820
Called Number            : mobile_phone
Source IP Address (Sig  ): CISCO_IP
Destn SIP Req Addr:Port  : PHONE_IP:5060
Destn SIP Resp Addr:Port : PHONE_IP:49153
Destination Name         : PHONE_IP

Jul 30 13:25:42.763: //1992896/D134DDBEB7DB/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): CISCO_IP
Source IP Port    (Media): 25012
Destn  IP Address (Media): PHONE_IP
Destn  IP Port    (Media): 27526
Orig Destn IP Address:Port (Media): [ - ]:0

Jul 30 13:25:42.763: //1992896/D134DDBEB7DB/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 487

VIP Super Bronze

These logs doesnt look

These logs doesnt look complete..Please use the instruction below to prperly collect logs.

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

Then..

<Enable debugs, then test again.>

debug ccsip messages

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


then do the ff:

terminal length 0
show logging

 

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

OK i try

OK i try

New Member

 i send all this commands in

 i send all this commands in config mode and aftre that send

term on

debug ccsip messages

and testing calls i dont have any messages on router console

VIP Super Bronze

yes you wont because its

yes you wont because its logged to router buffer..Follow the instruction.you need putty to collect the logs

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"
New Member

ok

ok

New Member

trace in attachment

trace in attachment

mobile_phone  - phone where call forwarding

6811 - phone where Call forward ENable

6820 - phone from Called to 6811

VIP Super Bronze

There is no sip calls in this

There is no sip calls in this trace. It just contains SIP REGISTER messages.

Please rate all useful posts "The essence of christianity is not the enthronement but the obliteration of self --William Barclay"

When sending the Invite out

When sending the Invite out to the other department, you're getting a 404 Not Found from them.

First attempt:

018708: Jul 30 18:11:11.245: //1994314/315DCB75B7F5/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:mobile_phone@172.31.5.2:5060 SIP/2.0
Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3C523CC17
Remote-Party-ID: "NAME" <sip:6820@CISCO_IP>;party=calling;screen=yes;privacy=off
From: "NAME" <sip:6820@CISCO_IP>;tag=613B3E8-2028
To: <sip:mobile_phone@172.31.5.2>
Date: Wed, 30 Jul 2014 14:11:11 GMT
Call-ID: 31603CAD-172A11E4-B7FBEC04-71085274@CISCO_IP
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0828230517-0388633060-3086347268-1896370804
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1406729471
Contact: <sip:6820@CISCO_IP:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Diversion: <sip:6811@CISCO_IP>;reason=unconditional;counter=1
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4795 2908 IN IP4 CISCO_IP
s=SIP Call
c=IN IP4 CISCO_IP
t=0 0
m=audio 25022 RTP/AVP 0 101 19
c=IN IP4 CISCO_IP
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

It sends the Invite again:

018711: Jul 30 18:11:11.741: //1994314/315DCB75B7F5/SIP/Msg/ccsipDisplayMsg:
Sent: 
INVITE sip:mobile_phone@172.31.5.2:5060 SIP/2.0
Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3C523CC17
Remote-Party-ID: "NAME" <sip:6820@CISCO_IP>;party=calling;screen=yes;privacy=off
From: "NAME" <sip:6820@CISCO_IP>;tag=613B3E8-2028
To: <sip:mobile_phone@172.31.5.2>
Date: Wed, 30 Jul 2014 14:11:11 GMT
Call-ID: 31603CAD-172A11E4-B7FBEC04-71085274@CISCO_IP
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 0828230517-0388633060-3086347268-1896370804
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1406729471
Contact: <sip:6820@CISCO_IP:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Diversion: <sip:6811@CISCO_IP>;reason=unconditional;counter=1
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 265

v=0
o=CiscoSystemsSIP-GW-UserAgent 4795 2908 IN IP4 CISCO_IP
s=SIP Call
c=IN IP4 CISCO_IP
t=0 0
m=audio 25022 RTP/AVP 0 101 19
c=IN IP4 CISCO_IP
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20

 

CME receives 100 Trying:

018712: Jul 30 18:11:11.761: //1994314/315DCB75B7F5/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 100 Trying
Date: Wed, 30 Jul 2014 14:11:21 GMT
From: "NAME" <sip:6820@CISCO_IP>;tag=613B3E8-2028
Allow-Events: presence
Content-Length: 0
To: <sip:mobile_phone@172.31.5.2>
Call-ID: 31603CAD-172A11E4-B7FBEC04-71085274@CISCO_IP
Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3C523CC17
CSeq: 101 INVITE

 

CME sends 404 Not Found:

018713: Jul 30 18:11:11.785: //1994314/315DCB75B7F5/SIP/Msg/ccsipDisplayMsg:
Received: 
SIP/2.0 404 Not Found
Reason: Q.850;cause=1
Date: Wed, 30 Jul 2014 14:11:21 GMT
From: "NAME" <sip:6820@CISCO_IP>;tag=613B3E8-2028
Allow-Events: presence
Content-Length: 0
To: <sip:mobile_phone@172.31.5.2>;tag=ad803d66-05af-4540-8ebe-481e90667f53-20224504
Call-ID: 31603CAD-172A11E4-B7FBEC04-71085274@CISCO_IP
Via: SIP/2.0/UDP CISCO_IP:5060;branch=z9hG4bK3C523CC17
CSeq: 101 INVITE

 

It may be because of the Diversion header in the Invite that the other side doesn't like or it may be because the number format is not correct.  You may want to compare to an Invite from a direct call that does work.

You could try removing the diversion header:

voice class sip-profiles 100
  request INVITE sip-header Diversion remove
voice service voip 
   sip
     sip-profiles 100

 

New Member

Thanks for your help. I try

Thanks for your help, Brian. I try remove diversion header and will notify you

New Member

Thanks for your help Brian

Thanks for your help Brian and Ayodeji

Brian you solution helps me. Thank you very much!

I can Forward ALL now to  08 ext numbers without rules

You said you can or can't

You said you can or can't forward all now?

New Member

My mistake . I can forward

My mistake . I can forward all
 

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