08-09-2011 09:06 AM - edited 03-16-2019 06:22 AM
Hello,
I have SIP configured Cisco 7960 phone which is receiving incoming SIP call and rejecting it. SIP debug messages are below telling something about "Call ID match not found". What does this mean?
[11:02:38:369711] sipTransportSendMessage: Closed a one-time UDP send channel handle = 5
[11:02:38:369716] SIPTaskProcessListEvent: cmd = 0x160200
[11:02:38:369716] SIPProcessUDPMessage: recv UDP message from <69.59.142.213>:<50195>, length=<354>, message=
[11:02:38:369717] ACK sip:artisticcheese@192.168.1.80:8061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 69.59.142.213:5060;branch=z9hG4bKadee01b823f34ae5b1226228b8d87925;rport
To: <sip:artisticcheese@sipsorcery.com>
From: "FRISCO TX" <sip:9721111111@66.54.140.46>;tag=1158419552
Call-ID: a46dd5fa862040f995618fee7ba64eec
CSeq: 1 ACK
Max-Forwards: 69
Content-Length: 0
[11:02:38:369719] SIPTaskProcessSIPMessage: Line filter: Determining destination line...
[11:02:38:369721] SIPTaskProcessSIPMessage: Line filter: Previous Call ID. Message not in reTx list.
[11:02:38:369721] SIPTaskProcessSIPMessage: Line filter: Call ID match not found: Rejecting.
Solved! Go to Solution.
08-10-2011 01:24 PM
The problem is the proxy port on line one.
The SIP INVITE is being sent to port 5060, the default SIP port number.
Via: SIP/2.0/UDP 69.59.142.213:5060
Contact: <69.59.142.213:5060>69.59.142.213:5060>
Notice what port calls are being sent to on line two.
Via: SIP/2.0/UDP 204.11.192.23:5080
So either change the proxy port on line one to 5060 or tell your SP to send calls to port 5070.
Let me know how it turns out.
08-09-2011 09:24 AM
Can we see the rest of this SIP call dialog? Is the Call-ID header, the tag of the From header, and the branch of the Via consistant since the INVITE was received?
08-09-2011 05:02 PM
Here is entire communication
debugs: timestamp
SIP Phone> debug ?
console-stall malloc-table sk-platform flash http timestamp timers dsp dsp-keepa
live vcm dtmf cpr-info task-socket xml-events xml-deck xml-vars xml-post lsm fsm
auth fim gsm cc cc-msg sip-task sip-state sip-messages sip-reg-state sip-trx cd
p dns sntp sntp-packet config
debugs: timestamp
SIP Phone> debug sip-task sip-state sip-messages sip-reg-state sip-trx
Enabling bug logging on this terminal - use 'tty mon 0' to disable
debugs: timestamp sip-task sip-state sip-messages sip-reg-state sip-trx
SIP Phone> [05:51:06:11435] SIPTaskProcessListEvent: cmd = 0x160200
[05:51:06:11435] SIPProcessUDPMessage: recv UDP message from <69.59.142.213>:<50
195>, length=<794>, message=
[05:51:06:11435] INVITE sip:artisticcheese@192.168.1.80:8061;transport=udp SIP/2
.0
Via: SIP/2.0/UDP 69.59.142.213:5060;branch=z9hG4bK4e958931769b4e32b97b84ecc457fe
f2;rport
To: <>>artisticcheese@sipsorcery.com>
From: "IRVING TX" <>;tag=1238637560>
Call-ID: 593fcd9141194c7d8898d534ccaddc08
CSeq: 1 INVITE
Contact: <69.59.142.213:5060>69.59.142.213:5060>
Max-Forwards: 69
User-Agent: www.sipsorcery.com
Content-Length: 334
Content-Type: application/sdp
v=0
o=root 16544 16544 IN IP4 66.54.140.46
s=session
c=IN IP4 66.54.140.46
t=0 0
m=audio 21864 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[05:51:06:11438] SIPTaskProcessSIPMessage: Line filter: Determining destination
line...
[05:51:06:11440] SIPTaskProcessSIPMessage: Line filter: Call ID match not found:
INVITE: free ccb index = 0.
[05:51:06:11441] Unknown address in Request URI
[05:51:06:11441] sipSPICheckRequest: Request URI Not Found
[05:51:06:11442] SIPTaskProcessSIPMessage: Error: sipSPICheckRequest() returned
error.
[05:51:06:11442] sipSPISendErrorResponse: Sending response 404...
[05:51:06:11444] sipRelDevCoupledMessageStore: Storing for reTx (cseq=1, method=
INVITE, to_tag=<>)
[05:51:06:11445] sipTransportSendMessage: Opened a one-time UDP send channel to
server <69.59.142.213>:<5060>, handle = 6 local port= 8061
[05:51:06:11446] sipTransportSendMessage:Sent SIP message to <69.59.142.213>:<50
60>, handle=<6>, length=<298>, message=
[05:51:06:11446] SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 69.59.142.213:5060;branch=z9hG4bK4e958931769b4e32b97b84ecc457fe
f2;rport
From: "IRVING TX" <>;tag=1238637560>
To: <>>artisticcheese@sipsorcery.com>
Call-ID: 593fcd9141194c7d8898d534ccaddc08
CSeq: 1 INVITE
Content-Length: 0
[05:51:06:11447] sipTransportSendMessage: Closed a one-time UDP send channel han
dle = 6
[05:51:06:11452] SIPTaskProcessListEvent: cmd = 0x160200
[05:51:06:11453] SIPProcessUDPMessage: recv UDP message from <69.59.142.213>:<50
195>, length=<354>, message=
[05:51:06:11453] ACK sip:artisticcheese@192.168.1.80:8061;transport=udp SIP/2.0
Via: SIP/2.0/UDP 69.59.142.213:5060;branch=z9hG4bK4e958931769b4e32b97b84ecc457fe
f2;rport
To: <>>artisticcheese@sipsorcery.com>
From: "IRVING TX" <>;tag=1238637560>
Call-ID: 593fcd9141194c7d8898d534ccaddc08
CSeq: 1 ACK
Max-Forwards: 69
Content-Length: 0
[05:51:06:11455] SIPTaskProcessSIPMessage: Line filter: Determining destination
line...
[05:51:06:11457] SIPTaskProcessSIPMessage: Line filter: Previous Call ID. Messag
e not in reTx list.
[05:51:06:11457] SIPTaskProcessSIPMessage: Line filter: Call ID match not found:
Rejecting.
08-09-2011 10:37 PM
Whats the firmware version of you 7960?
08-09-2011 11:59 PM
Cisco Systems, Inc. Copyright 2000-2005
Cisco IP phone MAC: 000b:beb2:5f93
Loadid: SW: P0S3-08-11-00 ARM: PAS3ARM1 Boot: PC030301 DSP: 4.0(5.0)[A0]
08-10-2011 02:09 AM
Does the actual handset rings when you receive inbound call?
Sent from Cisco Technical Support iPad App
08-10-2011 04:56 AM
No. It immediately gives busy signal back to caller.
08-10-2011 08:41 AM
Can send dial peer de bugs
Sent from Cisco Technical Support iPad App
08-10-2011 08:43 AM
? I don't understand. How do I fix it?
charbles2006 wrote:
Can send dial peer de bugs
Sent from Cisco Technical Support iPad App
08-10-2011 09:56 AM
Does your 7960 have a directory number assigned to it?
What are you using as your call processor or registrar server?
It looks like what is happening is your phone is receiving an INVITE to artisticcheese@sipsorcery.com but the phone doesn't have a line to attached the call to.
08-10-2011 09:58 AM
Entire configuration file is below. Failure happens when call comes to first number but rings fine if it comes to second one.
How do I make phone accept call on line 1 if it comes to "artisticcheese@sipsorcery.com"?
line1_displayname: "12063125984"
line1_name: "artisticcheese"
line1_authname: "artisticcheese"
line1_password: "mypass"
proxy1_address: "sipsorcery.com"
proxy1_port: 5070
line2_displayname: "17772702653"
line2_name: "17772702653"
line2_authname: "17772702653"
line2_password: "mypass"
proxy2_address: "callcentric.com"
proxy2_port: 5080
08-10-2011 10:10 AM
Can you send a debug for a call that works on the second line?
08-10-2011 12:55 PM
Here is log from successfull connection to line2
_STATE_RECV_INVITE_PROCEEDING <- E_CC_ALERTING
[15:52:34:29477] SIPSPISendInviteResponse: Sending response 180...
[15:52:34:29478] get_method_request_trx_index: Getting TRX for method(INVITE), s
ent = 0
[15:52:34:29478] get_method_request_trx_index: Got TRX(0) for recv method(INVITE
)
[15:52:34:29480] get_method_request_trx_index: Getting TRX for method(INVITE), s
ent = 0
[15:52:34:29480] get_method_request_trx_index: Got TRX(0) for recv method(INVITE
)
[15:52:34:29482] get_method_request_trx_index: Getting TRX for method(INVITE), s
ent = 0
[15:52:34:29483] get_method_request_trx_index: Got TRX(0) for recv method(INVITE
)
[15:52:34:29484] sipRelDevCoupledMessageStore: Storing for reTx (cseq=102, metho
d=INVITE, to_tag=<000bbeb25f93000879c60471-48abd6c0>)
[15:52:34:29486] sipTransportSendMessage: ccb <0>: config <204.11.192.23>:<5080>
- remote <204.11.192.23>:<5080>
[15:52:34:29487] sipTransportSendMessage: Got handle 3
[15:52:34:29487] sipTransportSendMessage: Opened a one-time UDP send channel to
server <204.11.192.23>:<5080>, handle = 8 local port= 5060
[15:52:34:29488] sipTransportSendMessage:Sent SIP message to <204.11.192.23>:<50
80>, handle=<8>, length=<645>, message=
[15:52:34:29488] SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 204.11.192.23:5080;branch=z9hG4bK-f3f7641485ff0585c0d78897f3b14
a33
From: <9729872279>;tag=as4f0cb9d29729872279>
To: <>>17772702653@ss.callcentric.com>;tag=000bbeb25f93000879c60471-48abd6c0
Call-ID: 039317b1746141a34cd428a61ec57e86@66.54.140.46
Date: Wed, 10 Aug 2011 19:52:34 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: <17772702653>17772702653>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: "17772702653" <>>17772702653@callcentric.com>;party=called;id
-type=subscriber;privacy=off;screen=yes
Content-Length: 0
[15:52:34:29490] sipTransportSendMessage: Closed a one-time UDP send channel han
dle = 8
[15:52:34:29490] LINE 0/2: sipTransportSendMessage : Stopping reTx ti
mer
[15:52:34:29490] CHANGE STATE: LINE 0/2: : St
ate change: SIP_STATE_RECV_INVITE_PROCEEDING -> SIP_STATE_RECV_INVITE_ALERTING
[15:52:41:30102] SIPTaskProcessListEvent: cmd = 0x0
[15:52:41:30102] LINE 7/2: --0x00071cdf-- : SIP_REG_STATE_RE
GISTERED <- E_SIP_REG_TMR_EXPIRE
[15:52:41:30102] LINE 7/2: sip_sm_call_cleanup : Cleaning up the
call
[15:52:41:30103] ccsip_register_send_msg: cmd=0x161c01
[15:52:41:30103] SIPTaskProcessListEvent: cmd = 0x161c01
[15:52:41:30104] LINE 7/2: --0x00071bc5-- : SIP_REG_STATE_ID
LE <- E_SIP_REG_REG_REQ
[15:52:41:30104] SIPSPISendRegister: Sending REGISTER...
[15:52:41:30105] get_next_request_trx_index: Getting next TRX index, sent = 1
[15:52:41:30106] get_next_request_trx_index: Got TRX(0) for sent req
[15:52:41:30106] get_last_request_trx_index: Getting last TRX index, sent = 1
[15:52:41:30106] get_last_request_trx_index: Got TRX(0) for sent req
[15:52:41:30108] get_last_request_trx_index: Getting last TRX index, sent = 1
[15:52:41:30108] get_last_request_trx_index: Got TRX(0) for sent req
[15:52:41:30110] get_last_request_trx_index: Getting last TRX index, sent = 1
[15:52:41:30111] get_last_request_trx_index: Got TRX(0) for sent req
[15:52:41:30112] sipTransportSendMessage: ccb <7>: config <204.11.192.23>:<5080>
- remote <204.11.192.23>:<5080>
[15:52:41:30113] sipTransportSendMessage: Got handle 3
[15:52:41:30113] sipTransportSendMessage: Opened a one-time UDP send channel to
server <204.11.192.23>:<5080>, handle = 8 local port= 5060
[15:52:41:30114] sipTransportSendMessage:Sent SIP message to <204.11.192.23>:<50
80>, handle=<8>, length=<570>, message=
[15:52:41:30114] REGISTER sip:callcentric.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.80:5060;branch=z9hG4bK55dc5072
From: <>>17772702653@callcentric.com>;tag=000bbeb25f93000928ecd406-4fbba26e
To: <>>17772702653@callcentric.com>
Call-ID: 000bbeb2-5f930003-228788c8-5067d40e@192.168.1.80
Max-Forwards: 70
Date: Wed, 10 Aug 2011 19:52:41 GMT
CSeq: 107 REGISTER
User-Agent: Cisco-CP7960G/8.0
Contact: <17772702653>;+sip.instance="17772702653>
uid:00000000-0000-0000-0000-000bbeb25f93>";+u.sip!model.ccm.cisco.com="7"
Content-Length: 0
Expires: 600
[15:52:41:30116] sipTransportSendMessage: Closed a one-time UDP send channel han
dle = 8
[15:52:41:30116] LINE 7/2: sipTransportSendMessage : Stopping reTx ti
mer
[15:52:41:30117] LINE 7/2: sipTransportSendMessage : Starting reTx ti
mer (500 msec)
[15:52:41:30117] LINE 7/2: Registration state change: SIP_REG_STATE_IDLE ---> SI
P_REG_STATE_REGISTERING
[15:52:41:30120] SIPTaskProcessListEvent: cmd = 0x160200
[15:52:41:30121] SIPProcessUDPMessage: recv UDP message from <204.11.192.23>:<55
Z315>, length=<461>, message=
[15:52:41:30121] SIP/2.0 200 Ok
v: SIP/2.0/UDP 192.168.1.80:5060;branch=z9hG4bK55dc5072;rport=5060;received=71.2
52.223.116
f: <>>17772702653@callcentric.com>;tag=000bbeb25f93000928ecd406-4fbba26e
t: <>>17772702653@callcentric.com>
i: 000bbeb2-5f930003-228788c8-5067d40e@192.168.1.80
CSeq: 107 REGISTER
m: <17772702653>;+sip.instance="
000000-0000-0000-0000-000bbeb25f93>";+u.sip!model.ccm.cisco.com=7;expires=65
l: 0
[15:52:41:30123] SIPTaskProcessSIPMessage: Line filter: Determining destination
line...
[15:52:41:30124] get_method_request_trx_index: Getting TRX for method(REGISTER),
sent = 1
[15:52:41:30124] get_method_request_trx_index: Got TRX(0) for sent method(REGIST
ER)
[15:52:41:30125] sip_sm_ccb_match_branch_cseq: Matched branch_id & CSeq
[15:52:41:30125] SIPTaskProcessSIPMessage: Line filter: Call ID match: Destinat
ion line = <7/2>.
[15:52:41:30126] SIPTaskProcessSIPMessage: Received SIP response.
[15:52:41:30127] get_method_request_trx_index: Getting TRX for method(REGISTER),
sent = 1
[15:52:41:30127] get_method_request_trx_index: Got TRX(0) for sent method(REGIST
ER)
[15:52:41:30128] sipSPICheckResponse: Response match: callid=000bbeb2-5f930003-2
28788c8-5067d40e@192.168.1.80, cseq=107, cseq_method=REGISTER
[15:52:41:30128] SIPTaskProcessSIPMessage: Stopping any outstanding reTx timers.
..
[15:52:41:30130] LINE 7/2: sip_sm_check_retx_timers : Stopping reTx ti
mer.
(callid=000bbeb2-5f930003-228788c8-5067d40e@192.168.1.80, cseq=107, cseq_method=
REGISTER)
[15:52:41:30131] SIPTaskProcessSIPMessage: Recv 2xx message.
[15:52:41:30131] LINE 7/2: --0x00072199-- : SIP_REG_STATE_RE
GISTERING <- E_SIP_REG_2xx
[15:52:41:30132] LINE 7/2: Registration state change: SIP_REG_STATE_REGISTERING
---> SIP_REG_STATE_REGISTERED
[15:52:41:30133] ccsip_handle_ev_2xx ui_set_sip_registration_state to TRUE 2
[15:52:41:30133] clean_method_request_trx: Removing TRX for method(REGISTER), se
nt = 1
[15:52:41:30133] clean_method_request_trx: Removed TRX(0) for method(REGISTER)
[15:52:41:30134] LINE 7/2: ccsip_get_exp_time_2xx : Using proxy expi
res value
[15:52:41:30135] LINE 7/2: ccsip_handle_ev_2xx : Starting expires
timer (60 sec)
[15:52:41:30135] sip_regmgr_trigger_fallback_monitor: Looking to trigger fallbac
k if any available
[15:52:43:30392] SIPTaskProcessListEvent: cmd = 0x160200
[15:52:43:30392] SIPProcessUDPMessage: recv UDP message from <204.11.192.23>:<55
315>, length=<331>, message=
[15:52:43:30392] CANCEL sip:17772702653@192.168.1.80:5060;transport=udp SIP/2.0
v: SIP/2.0/UDP 204.11.192.23:5080;branch=z9hG4bK-f3f7641485ff0585c0d78897f3b14a3
3
f: <9729872279>;tag=as4f0cb9d29729872279>
t: <>>17772702653@ss.callcentric.com>
i: 039317b1746141a34cd428a61ec57e86@66.54.140.46
CSeq: 102 CANCEL
Max-Forwards: 15
l: 0
08-10-2011 01:24 PM
The problem is the proxy port on line one.
The SIP INVITE is being sent to port 5060, the default SIP port number.
Via: SIP/2.0/UDP 69.59.142.213:5060
Contact: <69.59.142.213:5060>69.59.142.213:5060>
Notice what port calls are being sent to on line two.
Via: SIP/2.0/UDP 204.11.192.23:5080
So either change the proxy port on line one to 5060 or tell your SP to send calls to port 5070.
Let me know how it turns out.
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