cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
2404
Views
3
Helpful
24
Replies

CFW All not working for external Calls

Leonardo Santana
Spotlight
Spotlight

I have a CUCM 8.6 and a SIP TRUNK for external calls, when i forward extension A to a external number and i try to call extension A i receive a busy tone.

The outgoing and incoming calls are working and the call forward all for internal number are working.

Follow my config:

telephony-service

call-forward pattern .T

Thanks

Regards
Leonardo Santana

*** Rate All Helpful Responses***
1 Accepted Solution

Accepted Solutions

Eliminate Diversion header.

I think the same can be done with a voice translation-profile, eliminating redirect-target.

View solution in original post

24 Replies 24

paolo bevilacqua
Hall of Fame
Hall of Fame

Seem like you're running CME, not CM.

Possibly, the ITSP doesn't allow you to send a calling number that is not assigned to you.

You can take debug "ccsip message" with "term mon" to confirm.

I think yes, i ll try to put this command:

Router(config)#voice service voip

Router(conf-voi-serv)#no supplementary-service sip moved-temporarily

I will let you now

Thanks!

Regards
Leonardo Santana

*** Rate All Helpful Responses***

How can i confirm that the ITSP doesnt accept the caller ID?

Thanks

Regards
Leonardo Santana

*** Rate All Helpful Responses***

Doing what I've already indicated above.

Sorry Paolo is a CUCME i forgot the "E".

Thanks!

Regards
Leonardo Santana

*** Rate All Helpful Responses***

Hello Paolo i put the command

Router(config)#voice service voip

Router(conf-voi-serv)#no supplementary-service sip moved-temporarily

and didnt work

follow the debugs?

Jan 10 11:04:46.772: //17203/BDDA3C80B72A/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x319B9CA8

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 1148900510

Called Number            : 65263373

Source IP Address (Sig  ): My IP

Destn SIP Req Addr:Port  : SIP PROXY:5060

Destn SIP Resp Addr:Port :SIP PROXY

Destination Name         :SIP PROXY

GWFIVETEN01#

Jan 10 11:04:46.772: //17203/BDDA3C80B72A/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): My IP

Source IP Port    (Media): 16746

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

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

Jan 10 11:04:46.772: //17203/BDDA3C80B72A/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 57

Disconnect Cause (SIP)   : 403

Do you have any idea? Im sendind the main number as the calling, and the called number is on the right format.

Thanks

Regards
Leonardo Santana

*** Rate All Helpful Responses***

I have indicated above the debugs you should take.

Follow the output of the ccsip messages:

Sent:

INVITE sip:65263373@10.210.65.16:5060 SIP/2.0

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK48B8

Remote-Party-ID: <1148900510>;party=calling;screen=no;privacy=off

From: <1148900510>;tag=317C38-154E

To: <65263373>

Date: Tue, 10 Jan 2012 15:47:34 GMT

Call-ID: 3F7B7DAD-3AD911E1-804FDD04-43CEEA6E@10.213.26.98

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 1065057709-0987304417-2152455428-1137633902

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1326210454

Contact: <1148900510>

Diversion: <511>;privacy=off;reason=unconditional;counter=1;screen=no

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 355

v=0

o=CiscoSystemsSIP-GW-UserAgent 7789 6280 IN IP4 10.213.26.98

s=SIP Call

c=IN IP4 10.213.26.98

t=0 0

m=audio 28792 RTP/AVP 18 8 100 101 19

c=IN IP4 10.213.26.98

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

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 0-16

a=rtpmap:19 CN/8000

Jan 10 15:47:34.547: //19/3F7B7DAD804B/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

From: <1148900510>;tag=317C38-154E

To: <65263373>;tag=6521420521392012110134833

Call-ID: 3F7B7DAD-3AD911E1-804FDD04-43CEEA6E@10.213.26.98

CSeq: 101 INVITE

Server: CS2000_NGSS/9.0

Timestamp: 1326210454

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK48B8

Contact: <10.210.65.16:5060>

k: 100rel

Content-Length: 0

Jan 10 15:47:34.595: //19/3F7B7DAD804B/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 403 Forbidden

From: <1148900510>;tag=317C38-154E

To: <65263373>;tag=6521420521392012110134833

Call-ID: 3F7B7DAD-3AD911E1-804FDD04-43CEEA6E@10.213.26.98

CSeq: 101 INVITE

Server: CS2000_NGSS/9.0

k: 100rel

Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK48B8

Content-Length: 0

Jan 10 15:47:34.595: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

GWFIVETEN01#ACK sip:65263373@10.210.65.16:5060 SIP/2.0

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK48B8

From: <1148900510>;tag=317C38-154E

To: <65263373>;tag=6521420521392012110134833

Date: Tue, 10 Jan 2012 15:47:34 GMT

Call-ID: 3F7B7DAD-3AD911E1-804FDD04-43CEEA6E@10.213.26.98

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Thanks Paolo!

Regards
Leonardo Santana

*** Rate All Helpful Responses***

Now you can compare with a normal working call, and fidn the difference, likely calling number or other identify detail.

Hello Paolo i see that the calling number is the same, here the output of the debug of one call to the same nmer that i trying to set the call forward on the IP phone

GWFIVETEN01#

Jan 10 16:24:57.463: //34/7652D4918076/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:65263373@10.210.65.16:5060 SIP/2.0

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK1615CF

Remote-Party-ID: <1148900510>;party=calling;screen=no;privacy=off

From: <1148900510>;tag=53B5B4-258F

To: <65263373>

Date: Tue, 10 Jan 2012 16:24:57 GMT

Call-ID: 7860F316-3ADE11E1-807BDD04-43CEEA6E@10.213.26.98

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 1985139857-0987632097-2155273476-1137633902

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1326212697

Contact: <1148900510>

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 355

v=0

o=CiscoSystemsSIP-GW-UserAgent 8582 2611 IN IP4 10.213.26.98

s=SIP Call

c=IN IP4 10.213.26.98

t=0 0

m=audio 27420 RTP/AVP 18 8 100 101 19

c=IN IP4 10.213.26.98

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

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 0-16

a=rtpmap:19 CN/8000

Jan 10 16:24:57.487: //34/7652D4918076/SIP/Msg/ccsipDisplayMsg:

Received:

GWFIVETEN01#SIP/2.0 100 Trying

From: <1148900510>;tag=53B5B4-258F

To: <65263373>;tag=a8e71421489462012110142556

Call-ID: 7860F316-3ADE11E1-807BDD04-43CEEA6E@10.213.26.98

CSeq: 101 INVITE

Server: CS2000_NGSS/9.0

Timestamp: 1326212697

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK1615CF

Contact: <10.210.65.16:5060>

k: 100rel

Content-Length: 0

GWFIVETEN01#

Jan 10 16:24:59.455: %SM_INSTALL-6-INST_RBIP: ISM0/0 received msg: RBIP Registration Request

GWFIVETEN01#

Jan 10 16:25:01.655: //34/7652D4918076/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 183 Session Progress

From: <1148900510>;tag=53B5B4-258F

To: <65263373>;tag=a8e71421489462012110142556

Call-ID: 7860F316-3ADE11E1-807BDD04-43CEEA6E@10.213.26.98

CSeq: 101 INVITE

Server: CS2000_NGSS/9.0

Require: 100rel

Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK1615CF

RSeq: 57

Contact: <10.210.65.16:5060>

k: 100rel

c: application/sdp

Content-Length: 229

v=0

o=PVG 1326212750100 1326212750100 IN IP4 10.212.233.11

s=-

p=+1 6135555555

c=IN IP4 10.212.233.11

t=0 0

m=audio 54212 RTP/AVP 18 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=fmtp:18 annexb=no

Jan 10 16:25:01.655: //34/7652D4918076/SIP/Msg/ccsipDisplayMsg:

Sent:

PRACK sip:10.210.65.16:5060;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK171BA3

From: <1148900510>;tag=53B5B4-258F

To: <65263373>;tag=a8e71421489462012110142556

Date: Tue, 10 Jan 2012 16:24:57 GMT

Call-ID: 7860F316-3ADE11E1-807BDD04-43CEEA6E@10.213.26.98

CSeq: 102 PRACK

RAck: 57 101 INVITE

Allow-Events: telephone-event

Max-Forwards: 70

Content-Length: 0

Jan 10 16:25:01.671: //34/7652D4918076/SIP/Msg/ccsipDisplayMsg:

Received:

GWFIVETEN01#SIP/2.0 200 OK

From: <1148900510>;tag=53B5B4-258F

To: <65263373>;tag=a8e71421489462012110142556

Call-ID: 7860F316-3ADE11E1-807BDD04-43CEEA6E@10.213.26.98

CSeq: 102 PRACK

Server: CS2000_NGSS/9.0

k: 100rel

Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK171BA3

Content-Length: 0

GWFIVETEN01#

Jan 10 16:25:04.495: //34/7652D4918076/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

From: <1148900510>;tag=53B5B4-258F

To: <65263373>;tag=a8e71421489462012110142556

Call-ID: 7860F316-3ADE11E1-807BDD04-43CEEA6E@10.213.26.98

CSeq: 101 INVITE

Server: CS2000_NGSS/9.0

k: 100rel

Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK1615CF

Contact: <10.210.65.16:5060>

c: application/sdp

Content-Length: 229

v=0

o=PVG 1326212750100 1326212750100 IN IP4

GWFIVETEN01#10.212.233.11

s=-

p=+1 6135555555

c=IN IP4 10.212.233.11

t=0 0

m=audio 54212 RTP/AVP 18 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=fmtp:18 annexb=no

Jan 10 16:25:04.499: //34/7652D4918076/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:10.210.65.16:5060;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 10.213.26.98:5060;branch=z9hG4bK18D7

From: <1148900510>;tag=53B5B4-258F

To: <65263373>;tag=a8e71421489462012110142556

Date: Tue, 10 Jan 2012 16:24:57 GMT

Call-ID: 7860F316-3ADE11E1-807BDD04-43CEEA6E@10.213.26.98

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Jan 10 16:25:06.015: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

BYE sip:1148900510@10.213.26.98:5060 SIP/2.0

From: <65263373>;tag=a8e71421489462012110142556

To: <1148900510>;tag=53B5B4-258F

i: 7860F316-3ADE11E1-807BDD04-43CEEA6E@10.213.26.98

CSeq: 1 BYE

User-agent: CS2000_NGSS/9.0

Max-Forwards: 70

Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK

v: SIP/2.0/UDP SPO1CS2K:5060;maddr=10.210.65.16;branch=z9hG4bK-81db1a-fb3fe024-6a36b0ed

k: 100rel

l: 0

Jan 10 16:25:06.019: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

GWFIVETEN01#SIP/2.0 200 OK

Via: SIP/2.0/UDP SPO1CS2K:5060;maddr=10.210.65.16;branch=z9hG4bK-81db1a-fb3fe024-6a36b0ed;received=10.210.65.16

From: <65263373>;tag=a8e71421489462012110142556

To: <1148900510>;tag=53B5B4-258F

Date: Tue, 10 Jan 2012 16:25:06 GMT

Call-ID: 7860F316-3ADE11E1-807BDD04-43CEEA6E@10.213.26.98

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 BYE

Reason: Q.850;cause=16

P-RTP-Stat: PS=187,OS=3740,PR=400,OR=8000,PL=0,JI=0,LA=0,DU=1

Content-Length: 0

I see that this disconnect cause:

403 Forbidden

57

Bearer capability not authorized

Any idea?

Regards
Leonardo Santana

*** Rate All Helpful Responses***

The calls are identical beside Diversion header. The ITSP never replies bad bearer capability or other cause for "Forbidden"..

Does your ITSP allow to have multiple simultaneous calls ?

You can ask them why the forward call fail, or try to suppress the Diversion header via a number translation-profile, or SIP profile.

Yes is very strange to receive this forbidden.

Yes the ITSP allow multiple calls

What can modify in sip profile?

Thanks

Regards
Leonardo Santana

*** Rate All Helpful Responses***

Eliminate Diversion header.

I think the same can be done with a voice translation-profile, eliminating redirect-target.

sorry Paolo im not familiar with this

how can i eliminate the diversion header?

thanks!

Regards
Leonardo Santana

*** Rate All Helpful Responses***