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sip

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flexphil

flexphil

Programmer
Joined
Feb 3, 2007
Messages
29
Location
BE
SIP
Outgoing isn't working at all.
Line returns always busy.

Incoming -> in the monitor, I see that the line come's in, but the phone is not ringing.
In the incoming call route everything is ok, I think.

Any suggestions what to do?

Thanks
 
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How is your ARS Set.
Who are you using for SIP?

More info is needed for us to help on this.
What messaging are you getting to and from your ITSP?

Thanks

ACA - IPOffice implement
ACA - IP Telephony
CCNA - Passed at last
 
Upvote 0 Downvote
Phonext.com
Function code to ars 9N to N, main
Ars main N; to N"@82.146.123.83", line 3 (sip)

A little bit info from the monitor:

650931mS SIP Rx: UDP 82.146.123.83:5060 -> 192.168.2.10:5060

INVITE sip:992041111@192.168.2.10;transport=udp SIP/2.0
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
From: "+3292800155" <sip:+3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>
Contact: <sip:3292800155@81.201.82.27>
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 INVITE
User-Agent: Vox Callcontrol
Max-Forwards: 16
Date: Mon, 05 May 2008 11:04:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 11691 11691 IN IP4 81.201.82.27
s=session
c=IN IP4 81.201.82.27
t=0 0
m=audio 13166 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
650932mS SIP Trunk: 17:Rx
INVITE sip:992041111@192.168.2.10;transport=udp SIP/2.0
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
From: "+3292800155" <sip:+3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>
Contact: <sip:3292800155@81.201.82.27>
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 INVITE
User-Agent: Vox Callcontrol
Max-Forwards: 16
Date: Mon, 05 May 2008 11:04:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 11691 11691 IN IP4 81.201.82.27
s=session
c=IN IP4 81.201.82.27
t=0 0
m=audio 13166 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
650932mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget
650935mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1
650935mS SipDebugInfo: SIP: ProcessInbound Message
650935mS SipDebugInfo: Create Incoming EndPoint voip
650936mS StunInfo: Info: media ports will not be mapped with STUN.
650936mS SipDebugInfo: License, Valid 1, Available 1, Consumed 0
650936mS SipDebugInfo: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
650936mS SipDebugInfo: ExtractCallIdFrom Message: Call Id value 2dd423a500a9047b0063ce84334454ef@81.201.82.27
650937mS SipDebugInfo: ExtractCallerFromMessage: From Tag is as5081b9b7
650937mS SipDebugInfo: No Accept Header
650938mS SipDebugInfo: Sending code 100 to method INVITE
650938mS SipDebugInfo: SendSIPResponse, Number of Tag Count, 0
650938mS SipDebugInfo: Sip_sendToNetwork packet of length 481
650939mS SipDebugInfo: SIP Line (17): SendToTarget 52927b53, 5060
650939mS SIP Trunk: 17:Tx
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
From: "+3292800155" <sip:+3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>;tag=5c47f7892c7f4d34
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

650939mS SIP Tx: UDP 192.168.2.10:5060 -> 82.146.123.83:5060

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
From: "+3292800155" <sip:+3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>;tag=5c47f7892c7f4d34
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

650939mS SipDebugInfo: initialising mTxnContext
650940mS SipDebugInfo: Present call entered
650940mS SipDebugInfo: INVITE Received ep f5dc8488, dialog f5dc68f0
650940mS SipDebugInfo: ExtractRouteFromRecord, entered
650940mS SipDebugInfo: LR is On and route is Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
650940mS SipDebugInfo: *********************************************************
650940mS SipDebugInfo: State Transtion form Old State 0 to New state 10
650941mS SipDebugInfo: *********************************************************
650941mS SipDebugInfo: Present Call, no match (3238080216) from URI in To header.
650941mS CMMap: Mapper::AllocateCodec unable to allocate resource
650941mS SipDebugInfo: No Resource avaible, wait in queue
650941mS SipDebugInfo: Sending code 182 to method INVITE
650942mS SipDebugInfo: SendSIPResponse, Number of Tag Count, 1
650942mS SipDebugInfo: Sip_sendToNetwork packet of length 486
650942mS SipDebugInfo: SIP Line (17): SendToTarget 52927b53, 5060
650942mS SIP Trunk: 17:Tx
SIP/2.0 182 Call Queued
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
From: "+3292800155" <sip:+3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>;tag=5c47f7892c7f4d34
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

650943mS SIP Tx: UDP 192.168.2.10:5060 -> 82.146.123.83:5060

SIP/2.0 182 Call Queued
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
From: "+3292800155" <sip:+3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>;tag=5c47f7892c7f4d34
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

650943mS SipDebugInfo: *********************************************************
650943mS SipDebugInfo: State Transtion form Old State 10 to New state 42
650943mS SipDebugInfo: *********************************************************
650944mS SipDebugInfo: SIP Line (17): Freed Txn Key 2016
651176mS RES: Mon 5/5/2008 13:02:08 FreeMem=83852348(31) CMMsg=4 (4) Buff=200 966 1000 7214 5 Links=745
651743mS SIP Rx: UDP 82.146.123.83:5060 -> 192.168.2.10:5060

651744mS SIP Trunk: 17:Rx
651744mS SipDebugInfo: SIP Line (17): Non SIP packet, ignore
659960mS SIP Rx: UDP 82.146.123.83:5060 -> 192.168.2.10:5060

CANCEL sip:992041111@192.168.2.10;transport=udp SIP/2.0
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
From: "3292800155" <sip:3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 CANCEL
User-Agent: Vox Callcontrol
Max-Forwards: 16
Content-Length: 0

659960mS SIP Trunk: 17:Rx
CANCEL sip:992041111@192.168.2.10;transport=udp SIP/2.0
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
From: "3292800155" <sip:3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 CANCEL
User-Agent: Vox Callcontrol
Max-Forwards: 16
Content-Length: 0

659961mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget
659962mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1
659963mS SipDebugInfo: SIP: ProcessInbound Message
659963mS SipDebugInfo: Find End Point 2dd423a500a9047b0063ce84334454ef@81.201.82.27
659963mS SipDebugInfo: Process SIP request dialog f5dc68f0, method CANCEL in state 42
659963mS SipDebugInfo: Sending code 200 to method CANCEL
659964mS SipDebugInfo: SendSIPResponse, Number of Tag Count, 0
659964mS SipDebugInfo: Sip_sendToNetwork packet of length 427
659964mS SipDebugInfo: SIP Line (17): SendToTarget 52927b53, 5060
659964mS SIP Trunk: 17:Tx
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
From: "3292800155" <sip:3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>;tag=5c47f7892c7f4d34
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 CANCEL
Content-Length: 0

659965mS SIP Tx: UDP 192.168.2.10:5060 -> 82.146.123.83:5060

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
From: "3292800155" <sip:3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>;tag=5c47f7892c7f4d34
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 CANCEL
Content-Length: 0

659965mS SipDebugInfo: Sending code 487 to method INVITE
659965mS SipDebugInfo: SendSIPResponse, Number of Tag Count, 0
659966mS SipDebugInfo: Sip_sendToNetwork packet of length 493
659966mS SipDebugInfo: SIP Line (17): SendToTarget 52927b53, 5060
659966mS SIP Trunk: 17:Tx
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
From: "+3292800155" <sip:+3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>;tag=5c47f7892c7f4d34
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

659966mS SIP Tx: UDP 192.168.2.10:5060 -> 82.146.123.83:5060

SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
Via: SIP/2.0/UDP 81.201.82.27:5060;branch=z9hG4bK04a31e82;rport=5060
Record-Route: <sip:82.146.123.83;ftag=as5081b9b7;lr=on>
From: "+3292800155" <sip:+3292800155@81.201.82.27>;tag=as5081b9b7
To: <sip:3238080216@82.146.123.83>;tag=5c47f7892c7f4d34
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

659967mS SipDebugInfo: *********************************************************
659967mS SipDebugInfo: State Transtion form Old State 42 to New state 40
659967mS SipDebugInfo: *********************************************************
659968mS CMLineRx: v=0
CMReleaseComp
Line: type=IPLine 17 Call: lid=0 id=1020 in=2
IE CMIERespondingPartyName (228) +3292800155
IE CMIERespondingPartyNumber (230)(P:0 S:100 T:3 N:100 R:4) number=+3292800155@81.201.82.27
IE CMIEDeviceDetail (231) LOCALE=nlb HW=8 VER=4 class=CMDeviceSIPTrunk type=0 number=17 channel=0 rx_gain=32 tx_gain=32
659968mS SipDebugInfo: ~SipTrunkEndpoint
659968mS SipDebugInfo: SIPDialog destructor ... f5dc68f0
659968mS SipDebugInfo: Dialog destructor is deleting store d SIP message
659969mS CMTARGET: 0.1020.2 -1 BaseEP: ~CMTargetHandler
659969mS SipDebugInfo: SIP Line (17): Freed Txn Key 2016
660176mS RES: Mon 5/5/2008 13:02:17 FreeMem=83868948(41) CMMsg=4 (4) Buff=200 966 1000 7214 5 Links=774
660605mS SIP Rx: UDP 82.146.123.83:5060 -> 192.168.2.10:5060

ACK sip:992041111@192.168.2.10;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
From: "3292800155" <sip:3292800155@81.201.82.27>;tag=as5081b9b7
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
To: <sip:3238080216@82.146.123.83>;tag=5c47f7892c7f4d34
CSeq: 102 ACK
User-Agent: Sip EXpress router(0.9.6 (i386/freebsd))
Content-Length: 0

660605mS SIP Trunk: 17:Rx
ACK sip:992041111@192.168.2.10;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.146.123.83:5060;branch=z9hG4bK347f.f068d536.0
From: "3292800155" <sip:3292800155@81.201.82.27>;tag=as5081b9b7
Call-ID: 2dd423a500a9047b0063ce84334454ef@81.201.82.27
To: <sip:3238080216@82.146.123.83>;tag=5c47f7892c7f4d34
CSeq: 102 ACK
User-Agent: Sip EXpress router(0.9.6 (i386/freebsd))
Content-Length: 0

660605mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget
660607mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1
660607mS SipDebugInfo: SIP: ProcessInbound Message
660607mS SipDebugInfo: Create Incoming EndPoint voip
660608mS StunInfo: Info: media ports will not be mapped with STUN.
660608mS SipDebugInfo: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
660608mS SipDebugInfo: ExtractCallIdFrom Message: Call Id value 2dd423a500a9047b0063ce84334454ef@81.201.82.27
660608mS SipDebugInfo: ExtractCallerFromMessage: From Tag is as5081b9b7
660608mS SipDebugInfo: ExtractContactFromMessage: cannot get From Header 2012
660609mS SipDebugInfo: Incoming Method, Parsing Failed 9
660609mS SipDebugInfo: SIPDialog destructor ... f5dc68f0
660609mS SipDebugInfo: ~SipTrunkEndpoint
660609mS CMTARGET: 0.1021.2 -1 BaseEP: ~CMTargetHandler
660609mS SipDebugInfo: SIP Line (17): Cannot free Txn Key 2015
661852mS SIP Rx: UDP 82.146.123.83:5060 -> 192.168.2.10:5060

661852mS SIP Trunk: 17:Rx
661852mS SipDebugInfo: SIP Line (17): Non SIP packet, ignore
664965mS SipDebugInfo: Timer 10 callback
665178mS RES: Mon 5/5/2008 13:02:22 FreeMem=83878336(34) CMMsg=4 (4) Buff=200 966 1000 7214 5 Links=775
665607mS SipDebugInfo: Timer 9 callback
671959mS SIP Rx: UDP 82.146.123.83:5060 -> 192.168.2.10:5060

671959mS
 
Upvote 0 Downvote
For SIP operation you need a license for H.323 VCM channels and of course the SIP licenses, appearantly you do not have this H.323 license:
650941mS CMMap: Mapper::AllocateCodec unable to allocate resource
 
Upvote 0 Downvote
Does this not mean he has one;
SipDebugInfo: License, Valid 1, Available 1, Consumed 0

Would this be lack of VCM

ACA - IPOffice implement
ACA - IP Telephony
CCNA - Passed at last
 
Upvote 0 Downvote
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