01-09-2012 05:43 AM - edited 03-16-2019 08:53 AM
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
Solved! Go to Solution.
01-10-2012 10:05 AM
Eliminate Diversion header.
I think the same can be done with a voice translation-profile, eliminating redirect-target.
01-09-2012 06:04 AM
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.
01-09-2012 06:06 AM
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!
01-09-2012 06:07 AM
How can i confirm that the ITSP doesnt accept the caller ID?
Thanks
01-09-2012 06:08 AM
Doing what I've already indicated above.
01-09-2012 07:38 AM
Sorry Paolo is a CUCME i forgot the "E".
Thanks!
01-10-2012 05:59 AM
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
01-10-2012 06:37 AM
I have indicated above the debugs you should take.
01-10-2012 08:18 AM
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=off1148900510>
From: <1148900510>;tag=317C38-154E1148900510>
To: <65263373>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>1148900510>
Diversion: <511>;privacy=off;reason=unconditional;counter=1;screen=no511>
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=317C38-154E1148900510>
To: <65263373>;tag=652142052139201211013483365263373>
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>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=317C38-154E1148900510>
To: <65263373>;tag=652142052139201211013483365263373>
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=317C38-154E1148900510>
To: <65263373>;tag=652142052139201211013483365263373>
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!
01-10-2012 08:22 AM
Now you can compare with a normal working call, and fidn the difference, likely calling number or other identify detail.
01-10-2012 08:31 AM
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=off1148900510>
From: <1148900510>;tag=53B5B4-258F1148900510>
To: <65263373>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>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=53B5B4-258F1148900510>
To: <65263373>;tag=a8e7142148946201211014255665263373>
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>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=53B5B4-258F1148900510>
To: <65263373>;tag=a8e7142148946201211014255665263373>
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>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=53B5B4-258F1148900510>
To: <65263373>;tag=a8e7142148946201211014255665263373>
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=53B5B4-258F1148900510>
To: <65263373>;tag=a8e7142148946201211014255665263373>
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=53B5B4-258F1148900510>
To: <65263373>;tag=a8e7142148946201211014255665263373>
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>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=53B5B4-258F1148900510>
To: <65263373>;tag=a8e7142148946201211014255665263373>
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=a8e7142148946201211014255665263373>
To: <1148900510>;tag=53B5B4-258F1148900510>
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=a8e7142148946201211014255665263373>
To: <1148900510>;tag=53B5B4-258F1148900510>
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?
01-10-2012 08:51 AM
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.
01-10-2012 09:30 AM
Yes is very strange to receive this forbidden.
Yes the ITSP allow multiple calls
What can modify in sip profile?
Thanks
01-10-2012 10:05 AM
Eliminate Diversion header.
I think the same can be done with a voice translation-profile, eliminating redirect-target.
01-10-2012 11:20 AM
sorry Paolo im not familiar with this
how can i eliminate the diversion header?
thanks!
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