Sip CANCEL message sent too soon

dan.letkeman · ‎07-14-2011

Hello,

My sip provider is saying that my router is sending the cancel request too soon and not waiting long enouph for there system to respond. Here are the logs from the router when I try to make a call:

Jul 14 23:13:02.800: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 211.211.211.1:5060;branch=z9hG4bKE41B1;received=211.211.211.1

From: <sip:2225551234@did.voip.les.net>;tag=E2B693A4-C21

To: <sip:12225551234@did.voip.les.net>;tag=as7d976ff6

Call-ID: A5FA4E39-ADA511E0-B0AE8F35-C53231D0@211.211.211.1

CSeq: 101 INVITE

User-Agent: LES.NET.VoIP

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

Contact: <sip:12225551234@64.34.181.47>

Content-Type: application/sdp

Content-Length: 216

v=0

o=root 19801 19801 IN IP4 64.34.181.47

s=session

c=IN IP4 64.34.181.47

t=0 0

m=audio 13480 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

Jul 14 23:13:02.804: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

CANCEL sip:12225551234@did.voip.les.net:5060 SIP/2.0

Via: SIP/2.0/UDP 211.211.211.1:5060;branch=z9hG4bKE41B1

From: <sip:2225551234@did.voip.les.net>;tag=E2B693A4-C21

To: <sip:12225551234@did.voip.les.net>

Date: Thu, 14 Jul 2011 23:12:58 GMT

Call-ID: A5FA4E39-ADA511E0-B0AE8F35-C53231D0@211.211.211.1

CSeq: 101 CANCEL

Max-Forwards: 70

Timestamp: 1310685182

Reason: Q.850;cause=127

Content-Length: 0

Jul 14 23:13:02.856: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 211.211.211.1:5060;branch=z9hG4bKE41B1;received=211.211.211.1

From: <sip:2225551234@did.voip.les.net>;tag=E2B693A4-C21

To: <sip:12225551234@did.voip.les.net>;tag=as7d976ff6

Call-ID: A5FA4E39-ADA511E0-B0AE8F35-C53231D0@211.211.211.1

CSeq: 101 INVITE

User-Agent: LES.NET.VoIP

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

Content-Length: 0

Jul 14 23:13:02.856: //18945/002829CE3800/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x31670E60

State of The Call : STATE_DEAD

TCP Sockets Used : NO

Calling Number : 2225551234

Called Number : 12225551234

Source IP Address (Sig ): 211.211.211.1

Destn SIP Req Addr:Port : 64.34.181.47:5060

Destn SIP Resp Addr:Port : 64.34.181.47:5060

Destination Name : did.voip.les.net

Jul 14 23:13:02.860: //18945/002829CE3800/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 : 6

Dtmf-relay Payload : 101 (tx), 101 (rx)

Source IP Address (Media): 211.211.211.1

Source IP Port (Media): 0

Destn IP Address (Media): 64.34.181.47

Destn IP Port (Media): 13480

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

Jul 14 23:13:02.860: //18945/002829CE3800/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC) : 127

Disconnect Cause (SIP) : 487

Jul 14 23:13:02.860: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 211.211.211.1:5060;branch=z9hG4bKE41B1;received=211.211.211.1

From: <sip:2225551234@did.voip.les.net>;tag=E2B693A4-C21

To: <sip:12225551234@did.voip.les.net>;tag=as7d976ff6

Call-ID: A5FA4E39-ADA511E0-B0AE8F35-C53231D0@211.211.211.1

CSeq: 101 CANCEL

User-Agent: LES.NET.VoIP

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

Contact: <sip:12225551234@64.34.181.47>

Content-Length: 0

Jul 14 23:13:02.860: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

vr#ACK sip:12225551234@did.voip.les.net:5060 SIP/2.0

Via: SIP/2.0/UDP 211.211.211.1:5060;branch=z9hG4bKE41B1

From: <sip:2225551234@did.voip.les.net>;tag=E2B693A4-C21

To: <sip:12225551234@did.voip.les.net>;tag=as7d976ff6

Date: Thu, 14 Jul 2011 23:12:58 GMT

Call-ID: A5FA4E39-ADA511E0-B0AE8F35-C53231D0@211.211.211.1

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Is there a timer command that I need to configure to fix this?

________________________________________________________________________

My config:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

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

h323

h225 timeout setup 3

sip

redirect contact order best-match

!

sip-ua

credentials username 111111 password 7 123456 realm did.voip.les.net

authentication username 111111 password 7 123456 realm did.voip.les.net

no remote-party-id

retry invite 4

retry response 3

retry bye 2

retry cancel 2

retry register 5

timers register 250

registrar dns:did.voip.les.net expires 3600

sip-server dns:did.voip.les.net

no suspend-resume

!

The interesting thing is that using this same exact configuration on a different router works fine.....The one that works is a 2811 and the one that doesn't work is a 2921

Currently running 15.0(1)M6 on the 2921

Running 15.0(1)M5) on the 2811

Dan.

linuxchild · ‎07-15-2011

try to make debug ccsip error , you may have more info

dan.letkeman · ‎07-15-2011

vr#debug ccsip error

SIP Call error tracing is enabled

vr#

Jul 15 14:55:14.286: //37/00243E700300/SIP/Error/sipSPI_ipip_set_history_info_header:

Not SIP2SIP mode

Jul 15 14:55:14.294: //37/00243E700300/SIP/Error/sipSPI_ipip_set_history_info_header:

Not SIP2SIP mode

Jul 15 14:55:14.294: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:12225551234@did.voip.les.net:5060 SIP/2.0

Via: SIP/2.0/UDP 211.211.211.211:5060;branch=z9hG4bK2FA20

From: <>2043248910@did.voip.les.net>;tag=33524D8-1DAE

To: <>12225551234@did.voip.les.net>

Date: Fri, 15 Jul 2011 14:55:14 GMT

Call-ID: 47D0B7B5-AE2911E0-801DAA89-456D6B@211.211.211.211

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

Min-SE: 1800

Cisco-Guid: 0002375280-3511943650-0050394625-0168544441

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

Allow: INVITE,

vr#OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1310741714

Contact: <2043248910>

Call-Info: <211.211.211.211:5060>;method="NOTIFY;Event=telephone-event;Duration=2000

"

Expires: 180

Allow-Events: telephone-event

Content-Length: 0

Jul 15 14:55:14.362: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 211.211.211.211:5060;branch=z9hG4bK2FA20;received=211.211.211.211

From: <>2043248910@did.voip.les.net>;tag=33524D8-1DAE

To: <>12225551234@did.voip.les.net>

Call-ID: 47D0B7B5-AE2911E0-801DAA89-456D6B@211.211.211.211

CSeq: 101 INVITE

User-Agent: LES.NET.VoIP

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

Contact: <12225551234>

Content-Length: 0

Jul 15 14:55:18.438: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 211.211.211.211:5060;branch=z9hG4bK2FA20;received=211.211.211.211

From: <>2043248910@did.voip.les.net>;tag=33524D8-1DAE

To: <>12225551234@did.voip.les.net>;tag=as15d7502b

Call-ID: 47D0B7B5-AE2911E0-801DAA89-456D6B@211.211.211.211

CSeq: 101 INVITE

User-Agent: LES.NET.VoIP

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

Contact: <12225551234>

Content-Type: application/sdp

Content-Length: 216

v=0

o=root 14962 14962 IN IP4 64.34.181.47

s=session

c=IN IP4 64.34.181.47

t=0 0

m=audio 15300 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

Jul 15 14:55:18.438: //37/00243E700300/SIP/Error/sipSPICheckReliableProvStringtag: Un

able to access supported header values

SIP: Attribute mid, level 1 instance 1 not found.

Jul 15 14:55:18.442: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

CANCEL sip:12225551234@did.voip.les.net:5060 SIP/2.0

Via: SIP/2.0/UDP 211.211.211.211:5060;branch=z9hG4bK2FA20

From: <>2043248910@did.voip.les.net>;tag=33524D8-1DAE

To: <>12225551234@did.voip.les.net>

Date: Fri, 15 Jul 2011 14:55:14 GMT

Call-ID: 47D0B7B5-AE2911E0-801DAA89-456D6B@211.211.211.211

CSeq: 101 CANCEL

Max-Forwards: 70

Timestamp: 1310741718

Reason: Q.850;cause=127

Content-Length: 0

Jul 15 14:55:18.510: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/UDP 211.211.211.211:5060;branch=z9hG4bK2FA20;received=211.211.211.211

From: <>2043248910@did.voip.les.net>;tag=33524D8-1DAE

To: <>12225551234@did.voip.les.net>;tag=as15d7502b

Call-ID: 47D0B7B5-AE2911E0-801DAA89-456D6B@211.211.211.211

CSeq: 101 INVITE

User-Agent: LES.NET.VoIP

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

Content-Length: 0

Jul 15 14:55:18.510: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIRemoveBranchName: invalid ccb,

bName or branch list for sipSPIRemoveB

Jul 15 14:55:18.510: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 211.211.211.211:5060;branch=z9hG4bK2FA20;received=211.211.211.211

From: <>2043248910@did.voip.les.net>;tag=33524D8-1DAE

To: <>12225551234@did.voip.les.net>;tag=as15d7502b

Call-ID: 47D0B7B5-AE2911E0-801DAA89-456D6B@211.211.211.211

CSeq: 101 CANCEL

User-Agent: LES.NET.VoIP

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

Contact: <12225551234>

Content-Length: 0

Jul 15 14:55:18.510: //37/00243E700300/SIP/Error/act_dying_new_message: Received unex

pected response, dropping it

Jul 15 14:55:18.510: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:12225551234@did.voip.les.net:5060 SIP/2.0

Via: SIP/2.0/UDP 211.211.211.211:5060;branch=z9hG4bK2FA20

From: <>2043248910@did.voip.les.net>;tag=33524D8-1DAE

To: <>12225551234@did.voip.les.net>;tag=as15d7502b

Date: Fri, 15 Jul 2011 14:55:14 GMT

Call-ID: 47D0B7B5-AE2911E0-801DAA89-456D6B@211.211.211.211

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Looks like about the same information....

linuxchild · ‎07-15-2011

Hard to say , but I have no clue , do you have a firwall in the middel , I would check the log to see if any thing is blocked

Regards

Leo Salcie Tejeda · ‎08-24-2012

Hi,

I'm getting this same issue, did you find a solution?

Regards

MGCP, CCNA, CCNA Voice certified

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dan.letkeman · ‎08-27-2012

Yes, I forgot to add this dial-peer

dial-peer voice 901 voip

incoming called-number 8.T

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

I used 8 as the prefix for my sip trunk calls and the router that was terminating the sip trunk didn't know what to do with the call, after I added this it worked, but I don't remember the reason at the moment.