TG0069en-Ed13 SIP Maintenance R10.x and R11 2 de 2
TG0069en-Ed13 SIP Maintenance R10.x and R11 2 de 2
TG0069en-Ed13 SIP Maintenance R10.x and R11 2 de 2
The Call Control Half Com “link” is used for the SIP extension users (SEPLOS), it corresponds to the “CSIP”
function.
According to the declaration type of the SIP equipment on the OXE, the behavior will be different on the
SIPMOTOR side, and also on the Call Handling side.
The exchanges between the SIPMOTOR and the Call Handling are different according to this declaration.
When an issue appears in case of SIP equipment involved on the communication, it is important to check if
the problem is from the SIPMOTOR or from the Call Handling.
When a call is done, we can see on the motortrace the exchange between the SIPMOTOR to the Call
handling.
+------------------------------------------------------------+
| Message received SIP ----> UA (neqt : XXXX)
When traces are done on OXE to find the cause of the issue, it is important to link the call in the SIPMOTOR
traces and the Call Hanling traces. To do this check the “neqt” number (the neqt is 2250 in the following
examples)
In SIPMOTOR traces:
o For incoming call, the neqt is seen before the “display_ipc_out” message:
Ed. 13 79 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
- For outgoing call, the neqt is given on the “display_ipc_in” message from the Call
handling
For traces analysis, follow all the exhanges using this neqt. It is not possible to get more than one active call
using this “neqt”. When the call is released, this “neqt” is freed for another call.
Examples:
- [CCall::receiveRequest] INVITE: The SIPMOTOR has received a SIP request and
the request is an INVITE.
- [CTransaction::changeState]: The SIPMOTOR has changed the state of a
transaction.
- [getFromHeader]: the SIPMOTOR gets the information from the FROM header in
case of SIP incoming call.
- [isDomainFromGwExt]: the SIPMOTOR checks if the information from the domain
part of the FROM corresponds to an external SIP gateway.
The information “event” and “message” are in relation with the direction of the call and the SIP message:
- “event” is for the Call Handling.
- “message” is for the SIPMOTOR.
Ed. 13 80 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
The information between the [...] can more or less be understood. They can help to find the root cause of the
issue.
The “neqt” number is used to link the SIPMOTOR and Call Handling traces
o To find this Session reference for an incoming call, search for “[CCall::receiveRequest]
INVITE” after the INVITE received from the remote SIP equipment.
The transation reference, this value can be used to follow the transaction status evolution and to get
information about this transaction
o To find this transaction reference for an outgoing call, search for “STATE CHANGED TO
INITIAL” before the INVITE sent to the remote SIP equipment.
o To find this transaction reference for an incoming call, search for “STATE CHANGED TO
INITIAL” after the INVITE received from the remote SIP equipment.
o For one transaction, there is a pair of references. A “clone” reference is associated to the
main one: if the main one is 21be, the second reference is 21bf associated with the 200ok
receive or sent. This reference is seen with this message after the 200ok.
Ed. 13 81 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
The dialog reference, this value can be used to follow the dialog evolution and to get information
about this dialog
- On this example, the dialog reference is “158a”
- For one dialog, there is a pair of reference, a “clone” reference associated to the
main one, if the main one is 158a, the second reference is 158b associated with the
200ok receive or sent. This reference is seen with this message after the 200ok.
Mon May 28 15:21:08 2012 158b [CDialog::CDialog] look for the transaction #0, transaction key
= z9hG4bKca60f1097ab026913ca3bf56995162be
- For the dialog, the transaction reference is linked. The dialog “158a” is linked to the
transaction “21be”.
- There is the same link for the “clone” references.
Mon May 28 15:21:08 2012 158b [CDialog::onTransactionState(pTrans = 21bf, previousState =
Proceeding, currentState = Completed, reason = Final resp reception]
The Session reference, this one is unique for the complete call (from INVITE to the 200ok of the
BYE)
The Transaction references, the number of references depends of the call events (put on hold,
transfer, etc...)
o The main one is created when the INVITE is sent or received
o The other ones are created if an event is coming for the dialog associated (ACK, BYE,
REINVITE, REFER, etc...)
Ed. 13 82 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
A permanent link is done between the Dialog (main and clone) and the Transactions (main and clones). Here
is an example for an incoming call with 2 REINVITEs and a BYE at the end:
UAC . . . . . UAS
(1) Assignation a reference to the session, dialog and transaction
(SIP set) (Proxy)
| | (4) Creation of the clone dialog and the first clone transaction,
|(1) INVITE | associated to the clone dialog
|-------------------->|
|(2) 100 Trying | (5) First clone transaction terminated
|<--------------------|
|(3) 180 Ringing |
|<--------------------| (6) Creation of the second clone transaction for the first REINVITE,
|(4) 200 OK | associated to the clone dialog
|<--------------------|
|(5) ACK | (8) Second clone transaction terminated
|-------------------->|
|(6) INVITE | (9) Creation of the third clone transaction for the second
|-------------------->| REINIVTE, associated to the clone dialog
|(7) 200 OK |
|<--------------------|
|(8) ACK | (11) Third clone transaction terminated
|-------------------->| (12) Creation of a non-INVITE transaction (BYE) for the clone dialog
|(9) INVITE |
|-------------------->| (13) BYE transaction terminated, main transaction terminated,
|(10) 200 OK | session terminated and dialogs terminated
|<--------------------|
|(11) ACK |
|-------------------->|
|(12) BYE |
|-------------------->|
|(13) 200 OK |
|<--------------------|
12.9.1 Incoming SIP call using a SIP Trunk Group: SIPMOTOR point of view
v=0
o=- 3 2 IN IP4 135.118.226.39
s=Sip_Phone
c=IN IP4 135.118.226.39
t=0 0
m=audio 7888 RTP/AVP 8 18 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:A56A9738C0BC4CEF8087E10840231621
-------------------------------------------------
Ed. 13 83 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
The SIPMOTOR checks the Call-Id to know if this INVITE is an INVITE or a REINVITE.
Mon May 28 16:41:57 2012 1153 [CCall::getDialog] Confirmed Dialog is not found (ID =
;f6448c0c)
Mon May 28 16:41:57 2012 1153 [CCall::getDialog] Initial Dialog Server not found
When a transaction is linked to a dialog, the transaction changed from INITIAL to PROCEEDING.
In this case, the SIP equipment doesn’t send “Session timer” information because the value is 0 (updated :
0).
The SIPMOTOR makes the link between the dialog, transaction, the branch and the Cseq number.
Mon May 28 16:41:57 2012 156c [CDialog::addTransaction] added transaction 21a5 with branch
z9hG4bK-d87543-46534e582323f252-1--d87543-, with CSeq 1
The “branch” is a parameter added to the “via” to identify it. Regarding rfc3261, all the branch values must
start by “z9hG4bK”.
The CSeq is used to identify and to order a transaction, it consists of a sequence number and a method.
Ed. 13 84 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
The SIPMOTOR checks for which OXE equipment the call is from.
Mon May 28 16:41:57 2012 [isDomainFromGwExt] Host from request is : 172.27.142.53.
Mon May 28 16:41:57 2012 [isDomainFromGwExt] User from request is : 31024
Mon May 28 16:41:57 2012 [domain not from an External Gateway.
Mon May 28 16:41:57 2012 1153[CMotorCall::setFilterUsedMode] To be traced = 0
Mon May 28 16:41:57 2012 1153[CMotorCall::initOfUserType] values are reseted
Mon May 28 16:41:57 2012 [getFromHeader] displayName="PC_sip_device".
Mon May 28 16:41:57 2012 [getFromHeader] =31024@oxe-ov.alcatel.fr.
Mon May 28 16:41:57 2012 [getFromHeader] clirPresent=0.
Mon May 28 16:41:57 2012 [isAddrInDico] user=31024 host=oxe-ov.alcatel.fr
Mon May 28 16:41:57 2012 [isUserInDico] 31024@oxe-ov.alcatel.fr
Mon May 28 16:41:57 2012 [isUserInDico] found in the dictionnary.
Mon May 28 16:41:57 2012 [isAddrInDico] sip device station OK
-
The SIPMOTOR checks first if the domain part from the PAI, and of the FROM if no PAI, to
see if the call is for an external SIP gateway.
- Here, we can see that the call is from a SIP Device.
The SIPMOTOR checks for whom the call is done .
Mon May 28 16:41:57 2012 [isAddrInDico] user=31004 host=oxe-ov.alcatel.fr
Mon May 28 16:41:57 2012 [isUserInDico] 31004@oxe-ov.alcatel.fr
Mon May 28 16:41:57 2012 isUserInDico] NOT found in the dictionnary.
Mon May 28 16:41:57 2012 [isAddrInDico] other sip user
Here the call is for an “other sip user”, that means the call is for a non SIP user, corresponding to a legacy
set (IPtouch).
The SIPMOTOR checks the number of licenses available.
Here the number of licenses is 25, that means, 25 calls are possible on SIP using a SIP Trunk Group or
SEPLOS users.
Mon May 28 16:41:57 2012 The user is ipadd not in any Domain range return state as -1
Ed. 13 85 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
All the information about this call are sent to the Stand-By CPU.
Mon May 28 16:41:57 2012 SendToSipgwCpuSec: Message sent to the STAND-BY CPU
Mon May 28 16:41:57 2012 [receiveInviteMessage] send RemoteSdp to the StandBy.
Mon May 28 16:41:57 2012 SendToSipgwCpuSec: Message sent to the STAND-BY CPU
The information are sent to the Stand-By, like this, in case of bascul the SIP call will not be lost and known
on the new main CPU
Ed. 13 86 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
Mon May 28 16:41:57 2012 SEND MESSAGE TO NETWORK (135.118.226.39:25648 [UDP]) (BUFF LEN =
547)
----------------------utf8-----------------------
SIP/2.0 180 Ringing
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Contact: sip:oxe-ov.alcatel.fr
User-Agent: OmniPCX Enterprise R10.0 j1.410.45
To: "31004" <sip:31004@oxe-ov.alcatel.fr>;tag=15654dedb5658c165fbba7b0026e6ae9
From: "PC_sip_device" <sip:31024@oxe-ov.alcatel.fr>;tag=f6448c0c
Call-ID: ZWEwMGI4YjUxNjMyOWRlZmEyNWEzYThmNzI4NDUzMGM.
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.118.226.39:25648;received=135.118.226.39;branch=z9hG4bK-d87543-
46534e582323f252-1--d87543-;rport=25648
Content-Length: 0
“180 Ringing” is sent to the SIPMOTOR without SDP
-------------------------------------------------
A
For each SIP call event, a message is send to the Stand-By CPU.
The SIPMOTOR checks if the SDP given is compatible with the SDP received in the INVITE.
Ed. 13 87 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
Due to this, the dialog reference and transaction reference are changed (internal SIPMOTOR functionning).
Mon May 28 16:41:58 2012 156d [CDialog::CDialog] look for the transaction #0, transaction key
= z9hG4bK-d87543-46534e582323f252-1--d87543-
Mon May 28 16:41:58 2012 156d [CDialog::CDialog] copy the transaction #0, transaction key =
z9hG4bK-d87543-46534e582323f252-1--d87543-
Mon May 28 16:41:58 2012 21a6 [CTransaction::CTransaction] Transaction is cloned in 4 state
1338216118 -> Mon May 28 16:41:58 2012 SEND MESSAGE TO NETWORK (135.118.226.39:25648 [UDP])
(BUFF LEN = 974)
----------------------utf8-----------------------
SIP/2.0 200 OK
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Contact: sip:oxe-ov.alcatel.fr
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R10.0 j1.410.45
Session-Expires: 1800;refresher=uas
P-Asserted-Identity: "IPtouch 172.27.1" <sip:31004@oxe-ov.alcatel.fr;user=phone>
Content-Type: application/sdp
To: "31004" <sip:31004@oxe-ov.alcatel.fr>;tag=15654dedb5658c165fbba7b0026e6ae9
From: "PC_sip_device" <sip:31024@oxe-ov.alcatel.fr>;tag=f6448c0c
Call-ID: ZWEwMGI4YjUxNjMyOWRlZmEyNWEzYThmNzI4NDUzMGM.
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.118.226.39:25648;received=135.118.226.39;branch=z9hG4bK-d87543-
46534e582323f252-1--d87543-;rport=25648
Content-Length: 241
v=0
o=OXE 1338216117 1338216117 IN IP4 172.27.142.53
s=abs
c=IN IP4 172.27.142.64
t=0 0
m=audio 32514 RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=maxptime:40
a=rtpmap:101 telephone-event/8000
-------------------------------------------------
The 200ok sent to the remote SIP equipment is generated with information from the INVITE received and
from the 200ok answer from the Call Handling.
Mon May 28 16:41:58 2012 21a6 [CTransaction::startTimer] Timer G is started (delay = 500 ms)
Mon May 28 16:41:58 2012 21a6 [CTransaction::startTimer] Timer H is started (delay = 32000
ms)
Ed. 13 88 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
Mon May 28 16:41:59 2012 RECEIVE MESSAGE FROM NETWORK (135.118.226.39:25648 [UDP])
----------------------utf8-----------------------
ACK sip:oxe-ov.alcatel.fr SIP/2.0
Via: SIP/2.0/UDP 135.118.226.39:25648;branch=z9hG4bK-d87543-b00f692e5d3a246e-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:31024@135.118.226.39:25648>
To: "31004"<sip:31004@oxe-ov.alcatel.fr>;tag=15654dedb5658c165fbba7b0026e6ae9
From: "PC_sip_device"<sip:31024@oxe-ov.alcatel.fr>;tag=f6448c0c
Call-ID: ZWEwMGI4YjUxNjMyOWRlZmEyNWEzYThmNzI4NDUzMGM.
CSeq: 1 ACK
User-Agent: Sip Phone
Content-Length: 0
-------------------------------------------------
The SIPMOTOR changes the status of the transaction.
Mon May 28 16:41:59 2012 21a6 [CTransaction::changeState] STATE CHANGED TO TERMINATED
After call establishment, the call can be released by the OXE or by the remote SIP equipment.
The BYE is a new transaction for a SIP call, in that case, the transaction reference it is “21a7”, and the status
is “INITIAL”.
Mon May 28 16:42:00 2012 21a7 [CTransaction::startTimer] Timer E is started (delay = 500 ms)
Mon May 28 16:42:00 2012 21a7 [CTransaction::startTimer] Timer F is started (delay = 16000
ms)
- The 200ok of the BYE request is received from the remote SIP equipment.
- The Call Handling sends a message to the SIPMOTOR to release the “neqt” associated to
this SIP call
Mon May 28 16:42:00 2012 [display_ipc_in] ------------ Begin ---------------
Mon May 28 16:42:00 2012 neqt : 2250 Id : -1
Mon May 28 16:42:00 2012 SIP EQT RELEASED
Mon May 28 16:42:00 2012 [display_ipc_in] ------------- End ----------------
Ed. 13 90 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
The BYE is a new transaction for a SIP call. In that case, the transaction reference it is “21a7”, and the status
is “INITIAL”.
- The SIPMOTOR changes the transaction state.
Mon May 28 16:42:00 2012 21a7 [CTransaction::changeState] STATE CHANGED TO TRYING
Ed. 13 91 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
- The Call Handling sends a message to the SIPMOTOR to release the “neqt” associated to
this SIP call
Mon May 28 16:42:00 2012 [display_ipc_in] ------------ Begin ---------------
Mon May 28 16:42:00 2012 neqt : 2250 Id : -1
Mon May 28 16:42:00 2012 SIP EQT RELEASED
Mon May 28 16:42:00 2012 [display_ipc_in] ------------- End ----------------
12.9.2 Incoming SIP call using a SIP Trunk Group: Call Handling point of view
The call arrives on the SIPMOTOR, and sent to the Call Handling
(292779:000028) +------------------------------------------------------------+
(292779:000029) | Message received SIP ----> UA (neqt : 2250)
(292779:000030) | INVITE : 31004@oxe-ov.alcatel.fr:5060 ; user=name
(292779:000031) | From : <PC_sip_device> 31024@oxe-ov.alcatel.fr:5060 ; user=name
(292779:000032) | To : <"31004"> 31004@oxe-ov.alcatel.fr:5060 ; user=name
(292779:000033) +------------------------------------------------------------+
(292779:000034) | SDP :
(292779:000035) | @IP:port = 135.118.226.39:7888
(292779:000036) | ALGOS :
(292779:000037) | PCMA
(292779:000038) | G729
(292779:000039) | DTMF : 101
(292779:000040) | DIRECTION : SEND & RECEIVE
(292779:000041) | cac : false
(292779:000042) | Prack_Required: 0
(292779:000043) | Allow_UPDATE: 0
(292779:000044) | autoAnswer : false
(292779:000045) +------------------------------------------------------------+
All the information received on the Call handling are given by the SIPMOTOR, the SIPMOTOR has already
done an analysis and a treatment of these information.
Ed. 13 92 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
We can see the “neqt” used to make the link between the SIPMOTOR trace and Call Handling trace (here
2250)
When a call uses a SIP Trunk Group, the call is treated throught this SIP Trunk Group like a call on a
legacy T2 Trunk Group.
The Call Handling generates a SETUP message with the information given in the INVITE. The SETUP differs
if the Trunk Group is ISDN or ABCF.
___________________________________________________________________________
| (292779:000128) Concatenated-Physical-Event :
| long: 177 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0 term: 0 type a5
| tei: 0 <<<< message sent : SETUP [05] Call ref : 00 15
| SENDING COMPLETE
|______________________________________________________________________________
|
| IE:[04] BEARER_CAPABILITY (l=3) 80 90 a3
| IE:[18] CHANNEL (l=2) a0 90 -> T2 : No B channel
| IE:[1c] FACILITY (l=84)
| [91] Discriminator of supplementary service applications
| [aa] NFE (l=6):
| [80] Source Entity (l=1) End_PTNX
| [82] Destination Entity (l=1) End_PTNX
| [8b] Interpretation APDU (l=1): DISCARD (0)
| [a1] INVOKE (l=25):
| Invoke Ident. : 2ee0 (12000)
| OP: ECMA RO_CALLING_NAME (0)
| [80] Name presentation allowed (l=13) 'PC_sip_device'
| [a1] INVOKE (l=43):
| Invoke Ident. : 0001 (1)
| OP: ALCATEL RO_CLASSMARKS (1)
| [30] Sequence (l=30)
| [80] Feature identifier (l=5) 06 04 70 1f 20
| [82] Cug (l=1) 00
| [ab] Sequence of Project data (l=18)
| [30] Sequence (l=16)
| OP :RO_CLASSMARKS_SUPPLEMENTARY_INFO_1 (134623475)
| [30] Sequence (l=10)
| [80] Trunk group feature (l=5) 06 00 00 20 04
| [83] Current entity (l=1) 01
| IE:[6c] CALLING_NUMBER (l=7) -> 09 81 Num : 31024
| IE:[70] CALLED_NUMBER (l=6) -> 80 Num : 31004
| IE:[7d] HLC (l=2) 91 81
| [95] Locking shift. codeset : 5
| IE:[32] EI_PARTY_CATEGORY (l=1) -> EXTENSION (1)
| [9f] Non-locking shift. codeset : 7
| IE:[06] EI_IP_PAYLOADS (l=2) : (COMP/ECE/VAD) -> G711a/0/0 G729/0/0
| [97] Locking shift. codeset : 7
| IE:[0a] EI_RTP_INFO (l=30)
| -> stop_packet=0 stop_rtp=0 h323=0 wc=1 rf=0 udp=1 rqm=0
| -> Transm_Bande=1 detection_Q23=1 dtmf_payload=101
| -> Port RTP = 7888, IPv4 : 135. 118. 226. 39.
| -> Port RTCP SR = 7889, IPv4 : 135. 118. 226. 39.
| -> Port RTCP RR = 7889, IPv4 : 135. 118. 226. 39.
| -> Port Fax = 0, IPv4 : 0. 0. 0. 0.
|______________________________________________________________________________
When the SIP message is from the SIPMOTOR to the Call Handling, the direction is “message sent”.
Ed. 13 93 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
The Call Ref is identical for outgoing and incoming messages (here Call ref : 00 15).
______________________________________________________________________________
| (292779:000294) Concatenated-Physical-Event :
| long: 101 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0 term: 0 type a5
| tei: 0 >>>> message received : ALERT (01) Call ref : 00 15
|______________________________________________________________________________
|
.|| IE:[1c] FACILITY (l=64)
[91] Discriminator of supplementary service applications
| [aa] NFE (l=6):
| [80] Source Entity (l=1) End_PTNX
| [82] Destination Entity (l=1) End_PTNX
| [8b] Interpretation APDU (l=1): DISCARD (0)
| [a1] INVOKE (l=28):
| Invoke Ident. : 2ee1 (12001)
| OP: ECMA RO_CALLED_NAME (1)
| [80] Name presentation allowed (l=16) 'IPtouch 172.27.1'
| [a1] INVOKE (l=20):
| Invoke Ident. : 0001 (1)
| OP: ALCATEL RO_CLASSMARKS (1)
| [30] Sequence (l=7)
| [80] Feature identifier (l=5) 06 44 7e 1f 04
| IE:[1e] PROGRESS_ID (l=2) 80 88
| [95] Locking shift. codeset : 5
| IE:[32] EI_PARTY_CATEGORY (l=1) -> EXTENSION (1)
| [9f] Non-locking shift. codeset : 7
| IE:[06] EI_IP_PAYLOADS (l=1) -> G729 Ece 1 Vad 0
| [9f] Non-locking shift. codeset : 7
| IE:[0a] EI_RTP_INFO (l=2)
| -> stop_packet=0 stop_rtp=0 h323=0 wc=0 rf=0 udp=1 rqm=0
|
The -> Transm_Bande=1
ALERT detection_Q23=1
has no RTP information, dtmf_payload=101
because the SDP on 18x is not set to true.
|______________________________________________________________________________
The “ALERT” is transformed on a SIP message to the SIPMOTOR, but first the Call Handling select
the good “neqt” to send the message to the SIPMOTOR.
Ed. 13 94 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
_____________________________________________________________________________
| (292789:000511) Concatenated-Physical-Event :
| long: 134 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0 term: 0 type a5
| tei: 0 >>>> message received : CONNECT (07) Call ref : 00 15
|______________________________________________________________________________
|
| IE:[1c] FACILITY (l=64)
| [91] Discriminator of supplementary service applications
| [aa] NFE (l=6):
| [80] Source Entity (l=1) End_PTNX
| [82] Destination Entity (l=1) End_PTNX
| [8b] Interpretation APDU (l=1): DISCARD (0)
| [a1] INVOKE (l=28):
| Invoke Ident. : 2ee2 (12002)
| OP: ECMA RO_CONNECTED_NAME (2)
| [80] Name presentation allowed (l=16) 'IPtouch 172.27.1'
| [a1] INVOKE (l=20):
| Invoke Ident. : 0001 (1)
| OP: ALCATEL RO_CLASSMARKS (1)
| [30] Sequence (l=7)
| [80] Feature identifier (l=5) 06 44 7e 1f 04
| IE:[4c] CONNECTED_NUMBER (l=7) -> 00 81 Num : 31004
| [95] Locking shift. codeset : 5
| IE:[32] EI_PARTY_CATEGORY (l=1) -> EXTENSION (1)
| [9f] Non-locking shift. codeset : 7
| IE:[06] EI_IP_PAYLOADS (l=1) -> G729 Ece 1 Vad 0
| [9f] Non-locking shift. codeset : 7
| IE:[0a] EI_RTP_INFO (l=30)
| -> stop_packet=0 stop_rtp=0 h323=0 wc=0 rf=0 udp=1 rqm=0
| -> Transm_Bande=1 detection_Q23=1 dtmf_payload=101
| -> Port RTP = 32514, IPv4 : 172. 27. 142. 64.
| -> Port RTCP SR = 32515, IPv4 : 172. 27. 142. 64.
| -> Port RTCP RR = 32515, IPv4 : 172. 27. 142. 64.
| -> Port Fax = 0, IPv4 : 0. 0. 0. 0.
|______________________________________________________________________________
The “CONNECT” has RTP information. These RTP information are used to create the SDP.
The “CONNECT” is transformed to a SIP message towards the SIPMOTOR, but first the Call
Handling selects the good “neqt” to send the message to the SIPMOTOR.
The SIPMOTOR receives the ACK from the remote SIP equipment, and this message.
(292794:000580) +------------------------------------------------------------+
(292794:000581) | Message received SIP ----> UA (neqt : 2250)
(292794:000582) | ACK
(292794:000583) +------------------------------------------------------------+
Ed. 13 95 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
________________________________________________________________________
| (292794:000586) Concatenated-Physical-Event :
| long: 18 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0 term: 0 type a5
| tei: 0 <<<< message sent : CONNECT ACK (0f) Call ref : 00 15
|______________________________________________________________________________
After call establishment, the call can be released by the OXE or by the remote SIP equipment.
______________________________________________________________________________
| (292810:000672) Concatenated-Physical-Event :
| long: 23 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0 term: 0 type a5
| tei: 0 >>>> message received : DISCONNECT [45] Call ref : 00 15
|______________________________________________________________________________
|
| IE:[08] CAUSE (l=3) 80 90 80 -> [90] NORMAL CALL CLEARING
|______________________________________________________________________________
- The “DISCONNECT” is transformed to a SIP message towards the SIPMOTOR, but first the
Call Handling selects the good “neqt” to send the message to the SIPMOTOR.
(292810:000682) SIP : [send_to_motor] ipcSend resultat : 0 sur eqt : 2250
...
(292810:000684) +------------------------------------------------------------+
(292810:000685) | Message sent UA (neqt : 2250-0) ----> SIP
(292810:000686) | BYE
(292810:000687) +------------------------------------------------------------+
- Answer of the BYE received by the SIPMOTOR and transmited to the Call Handling.
(292811:000692) +------------------------------------------------------------+
(292811:000693) | Message received SIP ----> UA (neqt : 2250)
(292811:000694) | Successful 200
(292811:000695) | RELATIVE REQUEST : BYE
(292811:000696) | No SDP
(292811:000697) +------------------------------------------------------------+
- After the “REL COMP”, the call is completely ended on Call Handling side.
Ed. 13 96 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
According to the problem, more options can be used on the Call Handling trace, so that more information are
displayed. In the previous example, the minimum of options were set to see the exchanges between the
SIPMOTOR and the Call Handling.
It is important to understand the link between SIPMOTOR traces and Call Handling traces to make a
minimum of analysis before opening a Service Request.
12.9.3 Incoming SIP call in case of SIP extension: SIPMOTOR point of view
Tue Jun 26 08:03:05 2012 RECEIVE MESSAGE FROM NETWORK (135.118.226.21:61618 [UDP])
----------------------utf8-----------------------
INVITE sip:31004@oxe-ov.alcatel.fr SIP/2.0
Via: SIP/2.0/UDP 135.118.226.21:61618;branch=z9hG4bK-d87543-9c72747c0d38bb69-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:31023@135.118.226.21:61618>
To: "31004"<sip:31004@oxe-ov.alcatel.fr>
From: "PC_sip_extenstion"<sip:31023@oxe-ov.alcatel.fr>;tag=c850be7c
Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: SIP Phone
Content-Length: 317
v=0
o=- 5 2 IN IP4 135.118.226.21
s=SIP Phone
c=IN IP4 135.118.226.21
t=0 0
m=audio 46194 RTP/AVP 8 18 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
The information “RECEIVE MESSAGE FROM NETWORK
------------------------------------------------- (135.118.226.21:61618[UDP])” is important to
know that the call is an incoming one from the SIP equipment 135.118.226.21 in UDP.
The OXE checks the Call-Id to know if this INVITE is an INVITE or a REINVITE.
Tue Jun 26 08:03:05 2012 11ef [CCall::getDialog] Confirmed Dialog is not found (ID = ;c850be7c)
Tue Jun 26 08:03:05 2012 11ef [CCall::getDialog] Initial Dialog Server not found
Here it is an INVITE because the dialog is not found.
Here, the transaction reference is “210c” and the dialog reference is “15fd”.
Ed. 13 97 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
When a transaction is linked to a dialog, the transaction changed from INITIAL to PROCEEDING.
Tue Jun 26 08:03:05 2012 SEND MESSAGE TO NETWORK (135.118.226.21:61618 [UDP]) (BUFF LEN =
350)
----------------------utf8-----------------------
SIP/2.0 100 Trying
To: "31004" <sip:31004@oxe-ov.alcatel.fr>
From: "PC_sip_extenstion" <sip:31023@oxe-ov.alcatel.fr>;tag=c850be7c
Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.118.226.21:61618;received=135.118.226.21;branch=z9hG4bK-d87543-
9c72747c0d38bb69-1--d87543-;rport=61618
Content-Length: 0
-------------------------------------------------
The 100 Trying is generated by the SIPMOTOR.
In this case, the SIP equipment doesn’t send “Session timer” information because the value is 0 (updated :
0).
The SIPMOTOR makes the link between the transaction, the branch and the Cseq number.
Tue Jun 26 08:03:05 2012 15fd [CDialog::addTransaction] added transaction 210c with branch
z9hG4bK-d87543-9c72747c0d38bb69-1--d87543-, with CSeq 1
The “branch” is a parameter added to the “via” to identify it. Regarding rfc3261, all the branch values must
start with “z9hG4bK”.
The CSeq is used to identify and to order a transaction. It consists of a sequence number and a method.
The SIPMOTOR checks from which OXE equipment the call is.
Ed. 13 98 TG0069
OmniPCX Enterprise
TROUBLESHOOTING GUIDE No. 0069 Session Iniation Protocol (SIP)
Here the number of licenses is 25, that means, 25 calls are possible on SIP using a SIP Trunk Group or
SEPLOS users
The call is sent to the Call handling on neqt 2066, regarding the type of SIP equipment detected by the
SIPMOTOR, some information are added or not on this message.
All the information about this call are sent to the Stand-By CPU.
Tue Jun 26 08:03:05 2012 SendToSipgwCpuSec: Message sent to the STAND-BY CPU
Tue Jun 26 08:03:05 2012 [receiveInviteMessage] send RemoteSdp to the StandBy.
Tue Jun 26 08:03:05 2012 SendToSipgwCpuSec: Message sent to the STAND-BY CPU
The information are sent to the Stand-By so that in case of bascul the SIP call will not be lost on the new
main CPU
A “100 Trying” is sent by the Call Handling , but ignored by the SIPMOTOR.
This 100 Trying generated by the Call Handling is used to assign a “session” number for this call on the Call
Handling side, but not used by the SIPMOTOR
A “180 Ringing” is sent by the Call Handling with SDP, for the moment, on a 18X message, the Call Handling
will put everytime a SDP, no possibility to disable it.
The SIPMOTOR checks if the SDP given is compatible with the SDP received in the INVITE.
1340690585 -> Tue Jun 26 08:03:05 2012 11ef[CMotorCall::makeResponseSdp] Audio media.
Tue Jun 26 08:03:05 2012 11ef[CMotorCall::appendAudioAttributToMedia] Direction: 0.
Tue Jun 26 08:03:05 2012 11ef[CMotorCall::appendAudioAttributToMedia] format 101
Tue Jun 26 08:03:05 2012 11ef[CMotorCall::makeResponseSdp] fromSdp.getMediaDesciprionCount :1
Tue Jun 26 08:03:05 2012 [sameCodec] accepted Format : 18.
Tue Jun 26 08:03:05 2012 [sameCodec] requested Format : 8.
Tue Jun 26 08:03:05 2012 [sameCodec] requested Format : 18.
Tue Jun 26 08:03:05 2012 [sameCodec] same Format.
Tue Jun 26 08:03:05 2012 11ef[CMotorCall::mediaAccepted] Media accepted: m=audio 32584
RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
The codecs from the INVITE were 8 and 18 and the answer contains 18. In that case the call is accepted by
SIPMOTOR for SDP point of view.
v=0
o=OXE 1340690585 1340690585 IN IP4 172.27.141.151
s=abs
c=IN IP4 172.27.143.131
t=0 0
m=audio 32584 RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=maxptime:40
For each
a=rtpmap:101 telephone-event/8000
SIP call event, a message is sent to the Stand-By CPU.
-------------------------------------------------
Tue Jun 26 08:03:05 2012 [receiveInformationalEvent] UpdateContext send on the StandBy.
The SIPMOTOR checks if the SDP given is compatible with the SDP received in the INVITE.
The codecs from the INVITE were 8 and 18. The answer contains 18. In that case the call is accepted by
SIPMOTOR for SDP point of view.
Due to this, the dialog reference and transaction reference are changed (internal SIPMOTOR functionning).
Tue Jun 26 08:03:08 2012 15fe [CDialog::CDialog] look for the transaction #0, transaction key = z9hG4bK-
d87543-9c72747c0d38bb69-1--d87543-
Tue Jun 26 08:03:08 2012 15fe [CDialog::CDialog] copy the transaction #0, transaction key = z9hG4bK-
d87543-9c72747c0d38bb69-1--d87543-
Tue Jun 26 08:03:08 2012 210d [CTransaction::CTransaction] Transaction is cloned in 4 state
Tue Jun 26 08:03:08 2012 SEND MESSAGE TO NETWORK (135.118.226.21:61618 [UDP]) (BUFF LEN = 984)
----------------------utf8-----------------------
SIP/2.0 200 OK
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Contact: sip:oxe-ov.alcatel.fr
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R10.0 j1.410.45
Session-Expires: 1800;refresher=uas
P-Asserted-Identity: "IPtouch 172.27.142.64" <sip:31004@oxe-ov.alcatel.fr;user=phone>
Content-Type: application/sdp
To: "31004" <sip:31004@oxe-ov.alcatel.fr>;tag=05b5888d18d4e78f3554a55dadeefb08
From: "PC_sip_extenstion" <sip:31023@oxe-ov.alcatel.fr>;tag=c850be7c
Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.118.226.21:61618;received=135.118.226.21;branch=z9hG4bK-d87543-
9c72747c0d38bb69-1--d87543-;rport=61618
Content-Length: 242
v=0
o=OXE 1340690585 1340690586 IN IP4 172.27.141.151
s=abs
c=IN IP4 172.27.142.64
t=0 0
m=audio 32514 RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=maxptime:40
a=rtpmap:101 telephone-event/8000
-------------------------------------------------
The 200ok sent to the remote SIP equipment is generated with information from the INVITE received and
from the 200ok answer from the Call Handling.
Tue Jun 26 08:03:08 2012 RECEIVE MESSAGE FROM NETWORK (135.118.226.21:61618 [UDP])
----------------------utf8-----------------------
ACK sip:oxe-ov.alcatel.fr SIP/2.0
Via: SIP/2.0/UDP 135.118.226.21:61618;branch=z9hG4bK-d87543-cc14ac1776189458-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:31023@135.118.226.21:61618>
To: "31004"<sip:31004@oxe-ov.alcatel.fr>;tag=05b5888d18d4e78f3554a55dadeefb08
From: "PC_sip_extenstion"<sip:31023@oxe-ov.alcatel.fr>;tag=c850be7c
Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.
CSeq: 1 ACK
User-Agent: SIP Phone
Content-Length: 0
-------------------------------------------------
Tue Jun 26 08:03:08 2012 15fe [CDialog::receiveAckRequest] the INVITE request is terminated
After call establishment, the call can be released by the OXE or by the remote SIP equipment.
The BYE is a new transaction for a SIP call, in that case, the transaction reference it is “2110”, and the status
is “INITIAL”.
Tue Jun 26 08:03:10 2012 2110 [CTransaction::startTimer] Timer E is started (delay = 500 ms)
Tue Jun 26 08:03:10 2012 2110 [CTransaction::startTimer] Timer F is started (delay = 16000 ms)
- The 200ok of the BYE request is received from the remote SIP equipment.
Tue Jun 26 08:03:10 2012 RECEIVE MESSAGE FROM NETWORK (135.118.226.21:61618 [UDP])
----------------------utf8-----------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.27.141.151;branch=z9hG4bK2385fb34fcefc38c24fa6848df37e986
Contact: <sip:31023@135.118.226.21:61618>
To: <sip:31023@oxe-ov.alcatel.fr>;tag=c850be7c
From: "31004"<sip:31004@oxe-ov.alcatel.fr>;tag=05b5888d18d4e78f3554a55dadeefb08
Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.
CSeq: 716266225 BYE
User-Agent: SIP Phone
Content-Length: 0
-------------------------------------------------
- The SIPMOTOR changes this transaction state.
Tue Jun 26 08:03:10 2012 2110 [CTransaction::changeState] STATE CHANGED TO COMPLETED
- The Call Handling sent a message to the SIPMOTOR to release the “neqt” associated to
this SIP call
Tue Jun 26 08:03:10 2012 [display_ipc_in] ------------ Begin ---------------
Tue Jun 26 08:03:10 2012 neqt : 2066 Id : 1
Tue Jun 26 08:03:10 2012 SIP EQT RELEASED
Tue Jun 26 08:03:10 2012 [display_ipc_in] ------------- End ----------------
The BYE is a new transaction for a SIP call, in that case, the transaction reference it is “21a7”, and the status
is “INITIAL”.
- The SIPMOTOR changes the transaction state.
Tue Jun 26 08:03:10 2012 2110 [CTransaction::changeState] STATE CHANGED TO TRYING
- The SIPMOTOR sends the 200 ok of the BYE to the remote SIP equipment.
Tue Jun 26 08:03:10 2012 SEND MESSAGE TO NETWORK (135.118.226.39:25648 [UDP]) (BUFF LEN =
546)
----------------------utf8-----------------------
SIP/2.0 200 OK
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R10.0 j1.410.45
To: "31004"<sip:31004@oxe-ov.alcatel.fr>;tag=efa4b05316a486724541975cb22707d1
From: "PC_sip_extenstion"<sip:31023@oxe-ov.alcatel.fr>;tag=c55fb830
Call-ID: MzIwMmRjNGI3YTE3ZjkwZTE0ODE4Y2IzZGU1ZTdjZDM.
CSeq: 2 BYE
Via: SIP/2.0/UDP 135.118.226.39:25648;received=135.118.226.39;branch=z9hG4bK-d87543-
cf501c2f3311d050-1--d87543-;rport=25648
Content-Length: 0
-------------------------------------------------
- The SIPMOTOR changes the transaction state.
Tue Jun 26 08:03:10 2012 2110 [CTransaction::changeState] STATE CHANGED TO COMPLETED
- The Call Handling sends a message to the SIPMOTOR to release the “neqt” associated to
this SIP call
Tue Jun 26 08:03:10 2012 [display_ipc_in] ------------ Begin ---------------
Tue Jun 26 08:03:10 2012 neqt : 2266 Id : -1
Tue Jun 26 08:03:10 2012 SIP EQT RELEASED
Tue Jun 26 08:03:10 2012 [display_ipc_in] ------------- End ----------------
12.9.4 Incoming SIP call in case of SIP extension: Call Handling point of view
The call arrives on the SIPMOTOR, and sending to the Call Handling
(600095:000062) CSIP @@@@@@@@@@@@@@@@@@@@@@@@@@@@@ 02066 activated @@@@@@@@@@@@@@@@@@@@@@@@@@@@@@
(600095:000063) CSIP_receiveSipMsg
(600095:000064) +------------------------------------------------------------+
(600095:000065) | Message received SIP ----> UA (neqt : 2066)
(600095:000066) | INVITE : 31004@oxe-ov.alcatel.fr:5060 ; user=name
(600095:000067) | From : <PC_sip_extenstion> 31023@oxe-ov.alcatel.fr:5060 ; user=name
(600095:000068) | To : <"31004"> 31004@oxe-ov.alcatel.fr:5060 ; user=name
(600096:000069) +------------------------------------------------------------+
(600096:000070) | SDP :
(600096:000071) | @IP:port = 135.118.226.21:46194
(600096:000072) | ALGOS :
(600096:000073) | PCMA
(600096:000074) | G729
(600096:000075) | DTMF : 101
(600096:000076) | DIRECTION : SEND & RECEIVE
(600096:000077) | cac : false
(600096:000078) | Prack_Required: 0
(600096:000079) | Allow_UPDATE: 0
(600096:000080) | autoAnswer : false
(600096:000081) +------------------------------------------------------------+
(600096:000082) ..activeChId 0 featureList START_CALL
...
In case of SIP Extension, the call Handling treatment for the call starts by the message “CSIP”, for SIP
extension point of view.
In the first line, the information “02066 activated” is used to inform that the Call Handling starts the treatment
of the SIP extension with the neqt 2066.
The Call Handling checks if a session is already opened for this SIP extension user.
(600096:000087) ..CSIPMsgSipInvite::getSession
(600096:000088) ....CSIP_getSessionFromRequestURI
(600096:000089) ......Didn't retrieve session for requestUri 31004
(600096:000090) ....CSIP_getFreeSession
(600096:000091) ......Got free session 1 for ChId 80 CSIP_INVITE_WAIT_STATUS_CH_ID
In that case, no session opened, the Call Handling assigns to this call the session number 1, for a second
call (if the first call is still up) the session will be 2, etc...
(600096:000094) ......CSIPSession#1ChId#80::sendSipInformational
(600096:000095) ........CSIPSession#1ChId#80::emitMsgToSIPMotor
(600096:000096) ..........SIP_INFORMATIONAL sent
(600096:000097) +------------------------------------------------------------+
(600096:000098) | Message sent UA (neqt : 2066-1) ----> SIP
(600096:000099) | Informational 100
(600096:000100) | RELATIVE REQUEST : INVITE
(600096:000101) | No SDP
(600096:000102) +------------------------------------------------------------+
This 100 Trying will not be taken in account by the SIPMOTOR, it is only used to start the session on the Call
handling side.
Getting the SDP information received
This 100 Trying will not be taken in account by the SIPMOTOR, it is used only to start the session on the Call
handling side.
Analysis of the SDP information
The Call handling gets the SDP infomation of the equipment for the RBT to generate the SDP of the
180
(600096:000195) CSIP_sendInfoCs : No call server informations authorization.
(600096:000198) chgt_local_rtp_info ptdemi->info.hinfo=0 ptdemi->neqt=2066
(600096:000199) chgt_local_rtp_info local.wc=0 distant.wc=0 before update
(600096:000200) chgt_local_rtp_info end local.wc=0 distant.wc=0
(600096:000201) CSIP_sendInfoCs : No call server informations authorization.
(600096:000203) CSIP_sendUpdateMsgFromCh call_id=1->1 neqt=2066->2066
state=SCREEN_DIAL_0_DIGIT->SCREEN_DIAL_DIGIT
(600096:000204) CSIP_constructDistantField UTF-8 SCREEN_DIAL_DIGIT key=1
(600096:000205) ""
(600096:000206) CSIP_constructOtherField UTF-8 SCREEN_DIAL_DIGIT key=1
(600096:000207) " PC" 31023
(600096:000208) CSIP_constructSdp Default case
(600096:000209) 172.27.143.131:32584 G_729_A DTMF=101 SIP_SENDRECV
(600096:000210) CSIP_constructOtherInfo clir=0 forward=0 autoAnswer=0
(600096:000211) ..CSIPMsgInFactory::makeMsgInCh
(600096:000212) ..new CSIPMsgChDialDigit at 0x54038ce8 - counter 1
(600096:000213) CSIP_sendUpdateMsgFromCh -> call CSIP_setFeatureList
(600096:000214) nulog_final: 0 typconv : 0 ptdemi->forwarded_neqph:-1
(600096:000215) CSIP_setFeatureList
(600096:000216) CSIP_sendInfoCs : No call server informations authorization.
Here, the IP address for the RBT is 172.27.143.131, and the port used is 32584 and the codec used is G729
(this information appears few times in the trace)
The 180 is generated by the Call Handling and sent to the SIPMOTOR.
(600096:000400) CSIP_receiveComAction
(600096:000401) ..activeChId 1 featureList --
(600096:000402) ..CSIP Queue CSIPMsgChCalledStatus
(600096:000403) ..CSIPMsgChCalledStatus::getSession
(600096:000404) ....CSIP_getSessionFromChId
(600096:000405) ......Retrieved session 1 for ChId 1
(600096:000406) ..CSIPMsgChCalledStatus::execute
(600096:000407) ....CSIPStateInviteWaitCalledStatus::doCSIPMsgChCalledStatus
(600096:000408) ......CSIP_findSessionInTransfer
(600096:000409) ........No session in transfer
(600096:000410) ......SUBSTATE_ACT_INFO1 0 (libre )
(600096:000411) ......CSIPSession#1ChId#1::setDistantSdp
(600096:000412) ........CSIPSession#1ChId#1::compareDistantSdp
(600096:000413) ..........Change 0.0.0.0:5060 DTMF=255 SIP_INACTIVE
(600096:000414) .......... -> 172.27.143.131:32584 G_729_A DTMF=101 SIP_SENDRECV
(600096:000415) ........CSIPSession#1ChId#1::resetIsSdpSentInInf
(600096:000416) ......CSIPSession#1ChId#1::sendSipInformational
(600096:000417) ........CSIPSession#1ChId#1::setIsSdpSentInInf
(600096:000418) ........CSIPSession#1ChId#1::emitMsgToSIPMotor
(600096:000419) ..........SIP_INFORMATIONAL sent
(600096:000420) +------------------------------------------------------------+
(600096:000421) | Message sent UA (neqt : 2066-1) ----> SIP
(600096:000422) | Informational 180
(600096:000423) | RELATIVE REQUEST : INVITE
(600096:000424) +------------------------------------------------------------+
(600096:000425) | SDP :
(600096:000426) | @IP:port = 172.27.143.131:32584
(600096:000427) | ALGOS :
(600096:000428) | G729
(600096:000429) | DTMF : 101
(600096:000430) | DIRECTION : SEND & RECEIVE
(600096:000431) +------------------------------------------------------------+
(600096:000432) ......CSIPSession#1ChId#1::changeState
(600096:000433) ........CSIPStateInviteWaitCalledStatus -> CSIPStateInvite180WaitConversation
The state of the session, for Call Handling point of view, is changed to
“CSIPStateInvite180WaitConversation”
The Call handling gets the SDP infomation of the equipment for the 200ok
Here, the IP address for the 200ok is 172.27.142.64, the used port is 32514 and the codec is G729. This
SDP corresponds to the IPtouch.
The 200ok is generated by the Call Handling and sent to the SIPMOTOR
(600121:000525) CSIP_receiveComAction
(600121:000526) ..activeChId 1 featureList START_CALL HOLD
(600121:000527) ..CSIP Queue CSIPMsgChConversation
(600121:000528) ..CSIPMsgChConversation::getSession
(600121:000529) ....CSIP_getSessionFromChId
(600121:000530) ......Retrieved session 1 for ChId 1
(600121:000531) ..CSIPMsgChConversation::execute
(600121:000532) ....CSIPStateInvite180WaitConversation::doCSIPMsgChConversation
(600121:000533) ......CSIPSession#1ChId#1::setDistantSdp
(600121:000534) ........CSIPSession#1ChId#1::compareDistantSdp
(600121:000535) ..........Change 172.27.143.131:32584 G_729_A DTMF=101 SIP_SENDRECV
(600121:000536) .......... -> 172.27.142.64:32514 G_729_A DTMF=101 SIP_SENDRECV
(600121:000537) ........CSIPSession#1ChId#1::resetIsSdpSentInInf
(600121:000538) ......CSIPSession#1ChId#1::setDistantClir
(600121:000539) ......CSIPSession#1ChId#1::setDistantName
(600121:000540) ......CSIPSession#1ChId#1::setDistantNumber
(600121:000541) ......CSIPSession#1ChId#1::sendSipSuccessful
(600121:000542) ........CSIPSession#1ChId#1::emitMsgToSIPMotor
(600121:000543) ..........SIP_SUCCESSFUL sent
(600121:000544) +------------------------------------------------------------+
(600121:000545) | Message sent UA (neqt : 2066-1) ----> SIP
(600121:000546) | Successful 200
(600121:000547) | RELATIVE REQUEST : INVITE
(600121:000548) +------------------------------------------------------------+
(600121:000549) | SDP :
(600121:000550) | @IP:port = 172.27.142.64:32514
(600121:000551) | ALGOS :
(600121:000552) | G729
(600121:000553) | DTMF : 101
(600121:000554) | DIRECTION : SEND & RECEIVE
(600121:000555) | AssertedAddress : <IPtouch 172.27.142.64> 31004@oxe-ov.alcatel.fr:5060
(600121:000556) | COLP
(600121:000557) +------------------------------------------------------------+
(600121:000558) ......CSIPSession#1ChId#1::changeState
(600121:000559) ........CSIPStateInvite180WaitConversation -> CSIPStateConnectedWaitAck
The state of the session, for Call Handling point of view, is changed to “CSIPStateConnectedWaitAck”.
The state of the session, for Call Handling point of view, is changed to “CSIPStateConnected”.
- The BYE is generated by the Call Handling and sent to the SIPMOTOR
(600143:000733) CSIP_receiveComAction
(600143:000734) ..activeChId 1 featureList HOLD
(600143:000735) ..CSIP Queue CSIPMsgChOnHook
(600143:000736) ..CSIPMsgChOnHook::getSession
(600143:000737) ....CSIP_getSessionFromChId
(600143:000738) ......Retrieved session 1 for ChId 1
(600143:000739) ..CSIPMsgChOnHook::execute
(600143:000740) ....CSIPStateConnected::doCSIPMsgChOnHook
(600143:000741) ......CSIPSession#1ChId#1::sendMsgToCh
(600143:000742) ........CSIP_HANG_UP
(600143:000743) ......CSIPSession#1ChId#1::sendSipBye
(600143:000744) ........CSIPSession#1ChId#1::emitMsgToSIPMotor
(600143:000745) ..........SIP_BYE sent
(600143:000746) +------------------------------------------------------------+
(600143:000747) | Message sent UA (neqt : 2066-1) ----> SIP
(600143:000748) | BYE
(600143:000749) +------------------------------------------------------------+
(600143:000750) ......CSIPSession#1ChId#1::changeState
(600143:000751) ........CSIPStateConnected -> CSIPStateByeWait200
The state of the session, for Call Handling point of view, is changed to “CSIPStateByeWait200”.
On the Call Handling, the SIP extension calls have a “session”, this is the evolution of the session state from
the INVITE to the 200ok of the BYE:
12.10.1 Forwards
The OXE is able to manage different types of forward. Then if an equipment performs a forward to a SIP
equipment, the SIP messages behavior will differ according to this forward type.
SIP phone C
(31026)
OmniPCX Enterprise
In this type of call the OXE sends an INVITE to C (for all types of fowards) . Here are the different types of
INVITE sent according to the declaration of the SIP equipment on OXE:
----------------------utf8-----------------------
INVITE sip:31026@172.27.141.210:27836;rinstance=e26a48b411982396 SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
Supported: histinfo,replaces,timer,path
User-Agent: OmniPCX Enterprise R10.0 j1.410.45
Session-Expires: 1800;refresher=uac
Min-SE: 900
Content-Type: application/sdp
To: "IPtouch 172.27.141" <sip:31000@oxe-ov.alcatel.fr;user=phone>
From: "IPtouch 172.27.142.64" <sip:31004@oxe-
ov.alcatel.fr;user=phone>;tag=fc0ad7be3c9267a849d2
789c08cf26d3
Contact: <sip:31004@oxe-ov.alcatel.fr;transport=UDP>
Call-ID: 3b392056e4729fbd0734266cac4106bf@172.27.141.151
CSeq: 960429378 INVITE
Via: SIP/2.0/UDP 172.27.141.151;branch=z9hG4bKc2893fd8925d9aa6704859e3fb78877a
Max-Forwards: 70
Content-Length: 240
In that case, the important information is the “TO” field containing the directory number of the user forwarded
to the SIP extension (31000 in that case). There’s no more information to indicate that the call is forwarded.
-------------------------------------------------
Most of the time the SIP equipment returns a 302 message to inform the proxy that the call is fowarded. This
message is immediate or after a delay according to the type of forward.
If the SIP equipment is a proxy, it is able to keep the call. In that case, 2 SIP legs are opened, one from the
OXE to the proxy, the second one from the proxy to the forwarded destination.
If the SIP equipment is declared as a SIP extension, the forwarding prefixes can be used on this equipment.
In that case no INVITE will be sent to the SIP equipment because the Call Handling knows that this user is
forwarded.
12.10.2 Transfer
To make a transfer, the OXE can use (receive and accept) different ways according to the call context:
OmniPCX Enterprise
----------------------utf8-----------------------
REFER sip:oxe-ov.alcatel.fr SIP/2.0
Via: SIP/2.0/UDP 172.27.141.210:63016;branch=z9hG4bK-d87543-5c3865307254f255-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:31026@172.27.141.210:63016>
To: "31004"<sip:31004@oxe-ov.alcatel.fr>;tag=171c87e6f9b80ed5f6819b411a72505c
From: "31026"<sip:31026@oxe-ov.alcatel.fr>;tag=15672359
Call-ID: ODFlNGNmY2JjNDgyOGEwNDRmYjhhY2NjODAxM2U2NWE.
CSeq: 3 REFER
User-Agent: SIP Phone
Refer-To: <sip:31000@oxe-ov.alcatel.fr>
Referred-By: <sip:31026@172.27.141.210:63016>
Content-Length: 0
-------------------------------------------------
On this REFER, the following information are present:
“Refer-To” contains the directory number of the transfer destination.
“Referred-By” contains the directory number of the user performing the transfer.
SIP/2.0 200 OK
-------------------------------------------------
The NOTIFY corresponds to the final state of the transfer. Here the NOTIFY has “200 Ok” at the end of the
message. In this example the transfer has be done by the OXE.
If the on NOTIFY, the information is 503 Unavailable, in that case, the transfer has failed. Some other
information can be present (488, 486, etc...) according to the failed cause.
-------------------------------------------------
At the end of the transfer the leg1 is closed by C and leg2 is closed by the SIPMOTOR, only the leg3 from
the A to D remains.
In some calls scenarios, the OXE will send or receive an UPDATE on Early Media (before dialog opened) to
change the SDP.
SIP phone C
(31026)
OmniPCX Enterprise
During the RINGING phase, the OXE will send an UPDATE (after sending the 180 RINGING) to C. The OXE
has to send a PRACK before sending the UPDATE, to make a Pre-Acknowledgment and receive a 200ok for
this PRACK.
After this, the OXE will be able to send the UPDATE.
- To send a PRACK the OXE needs a “Require: 100rel” on the 18X answer received:
Mon Jun 11 15:01:38 2012 RECEIVE MESSAGE FROM NETWORK (172.27.143.186:5060 [UDP])
----------------------utf8-----------------------
SIP/2.0 180 Ringing
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE, OPTIONS, UPDATE
Contact: sip:172.27.143.186
Require: 100rel
User-Agent: SIP Phone
To: <sip:31006@172.27.143.186;user=phone>;tag=d7758dbc7f49c9521d28e60ef312ab04
From: "IPtouch 172.27.1" <sip:31000@oxe-
ov.alcatel.fr;user=phone>;tag=0c835efa2e1bf86a90d0016a
Call-ID: d626cd71ab0eab5c0499c46fd5324a91@172.27.141.151
CSeq: 679245852 INVITE
Via: SIP/2.0/UDP 172.27.141.151;branch=z9hG4bK61c571ebc4b1f5e5ff9e122e7e8b4a06
RSeq: 1131790336
Content-Length: 0
- After receiving this “Require: 100rel”, the OXE generates the PRACK
-------------------------------------------------
Mon Jun 11 15:01:38 2012 SEND MESSAGE TO NETWORK (172.27.143.186:5060 [UDP]) (BUFF LEN = 514)
----------------------utf8-----------------------
PRACK sip:172.27.143.186 SIP/2.0
Supported: replaces,timer,path
User-Agent: OmniPCX Enterprise R10.0 j1.410.45
RAck: 1131790336 679245852 INVITE
To: <sip:32006@172.27.143.186;user=phone>;tag=d7758dbc7f49c9521d28e60ef312ab04
From: <sip:31000@oxe-ov.alcatel.fr;user=phone>;tag=0c835efa2e1bf86a90d0016a0389c18e
Call-ID: d626cd71ab0eab5c0499c46fd5324a91@172.27.141.151
CSeq: 679245853 PRACK
Via: SIP/2.0/UDP 172.27.141.151;branch=z9hG4bK8b757b21da861556184ff74e5f5aaca7
Max-Forwards: 70
Content-Length: 0
-------------------------------------------------
v=0
o=OXE 1339422663 1339422663 IN IP4 172.27.141.151
s=abs
c=IN IP4 172.27.142.64
t=0 0
m=audio 32514 RTP/AVP 18 97
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
-------------------------------------------------
The UAS receiving this UPDATE is able to use the connection point for the RTP flow
When you connect a SIP equipment, it is mandatory to check if this equipment is tested and validated by
Alcatel-Lucent
- The SIP equipments like faxs, sets, etc… are validated via the AAPP. The
Configuration procedures are available on BPWS.
- The SIP providers test the connection with OXE themselves. So if you want to
connect one SIP provider, check if this provider has done the interopability test. All
the configuration procedures are given by the providers and not by Alcatel-Lucent.
General Parameters
- DPNSS prefix (necessary for optimisation on call forward).
- System codec (G729, G723).
- Support of multi-algo should be set to false.
Netadmin
- Use of specific characters (& _ $ ...) is not allowed for the nodename.
- Activate internal name resolver in spatial redundancy topologies.
SIP Proxy
- By default, the SIP proxy is set with “SIP Digest” for the Minimal authentication method, but
there is no Realm managed, so it is necessary to disable the authentication (SIP None) or to
manage a Realm.
In case of SSH management, the SIP equiments must be managed as SIP gateway (choice 1).
On the OXE SIP incidents are generated on Call Handling side, thes incidents are linked to a SIP alarm (files
under /tmpd), here an example of SIP alarm generated:
In that situation, the OXE receives a “SUBSCRIBE” message, but is not able to answer it, because the
purpose of this “SUBSCRIBE” message is unknown by the OXE.
When this types of alarm are present on the OXE, remove the Subscription on the remote SIP equipment to
avoid the Alarm.
When lots of alarms like these ones are generated on OXE, they can cause a “crash” of the SIPMOTOR.
Alarm due to bad SIP call context not copied on Stand-By CPU:
In that situation, on the INVITE received, the VIA header is too long for the OXE and it is not able to send the
SIP “context” to the stand by CPU. The call is established, but in case of bascul, this will not be known by the
new main CPU.
When the Information is “receiveInviteEvent”, the Call Handling sends an INVITE to the SIPMOTOR, but due
to a lack of ressources or licenses the INVITE cannot be sent by the SIPMOTOR.
When the Information is “receiveInviteMessage”, the SIPMOTOR has received an INVITE but due to a lack
of ressources (channels on SIP Trunk Group, CAC, compressors, ...) or licenses, the SIPMOTOR cannot
send the INVITE to the Call Handling.
Alarm due to a request not for the SIP proxy of the OXE:
This alarm means that the SIPMOTOR receives a SIP request that’s not for it, and is not able to route it to
another SIP equipment. It’s necessary to make a SIPMOTOR traces to get the IP address of this SIP
equipment.
The SIPMOTOR is not able to send a SIP message to a SIP extension. Remove the fact to send this
message on the SIP extension phone cos.
The SIPMOTOR generates this alarm because it is not able to send a CANCEL message, because the
dialog is already opened. The Call Handling asks the SIPMOTOR to send a CANCEL, but the 200ok for this
INVITE transaction is already arrived.
The SIPMOTOR generates this alarm because it is not able to ACK an INVITE transaction, because the
transaction is already terminated. Open a SR for analysis.
This part is used to explain the general possible issues on the OXE, not for a specific equipment
12.11.3.1 SIPMOTOR
- Symptom: With the ‘ps -edf | grep sipmotor’ command, no processes are present
- Explanation: This is due to a bad configuration of the SIP on your OXE. For instance the SIP
Trunk group managed on the local SIP gateway is not a SIP Trunk Group.
- Solution: Manage the good configuration and a restart of the CPU is mandatory.
- Symptom: With the ‘ps -edf | grep sipmotor’ command, only 2 SIPMOTOR processes are
present
- Explanation: When a modification is done on the SIP Trunk Group associated to the local
SIP gateway, for instance to replace Mini SIP Trunk group by a SIP Trunk group, the OXE
needs do resize the memory space due to this modification (often after the first management
of the local SIP gateway)
- Symptom: SIPMOTOR is rejecting all the call by a 503 message, and with the tool
“sipdump”, the status of the SIPMOTOR is in “degraded” mode
- Explanation: This a protection for the SIPMOTOR, when there are too many SIP “instance”
in the SIPMOTOR, the SIPMOTOR switches in degraded mode to protect itself. When it has
this status, all the incoming SIP requests are rejected by a 503. This mechanism avoids the
application from being overwhelmed by the traffic.
- Solution: nothing can be done, the SIPMOTOR will disable this mode automaticaly due to
some internal timers and thresholds. However, check that all Remote Domain and SIP
Outbound Proxy addresses are correctly added on Trusted IP Addresses.
- Symptom: If a restart of the SIPMOTOR is performed, all the SIP call contexts are lost
- Explanation: The restart of the SIPMOTOR provides the loss of all the SIP contexts. If SIP
calls are established, the RTP flow is maintained. At the SIP point view the call is not
present anymore, which means that if the SIPMOTOR receives a BYE for a call, the BYE will
be answered by a “481 Call/Transaction Does Not Exist”, but the call will be stopped. Also if
you use the session timer (time to check if the call is still up for the SIP point of view) the call
will be cut by the OXE because the context is unknown by the SIPMOTOR
- Solution: This is a normal behaviour if the restart is done manually. If the SIPMOTOR
automatically restarts a SR must be opened for analysis.
- Explanation: When the SIP is managed on the OXE, the SIPMOTOR processes uses
memory space. When the traffic is going up, the used memory space is increasing. When
the traffic rate is going down, the memory space used is decreasing.
Now, if when the traffic rate is going down, the memory space used doesn’t decrease
correctly, and if day after day, even if there is no traffic, the used memory is growing, the
SIPMOTOR will finally crash. In such case, the SIPMOTOR has problems to “delete” some
SIP contexts from its memory. After accumulation of the not deleted SIP contexts, the
SIPMOTOR cannot work properly and crashes.
- Action to do:
- Solution: A restart of the SIPMOTOR can be done and due to this, all the SIP contexts are
deleted. The problem will be solved but only for a time, if the root cause is not found, the
problem will be back again. Open a SR for analysis.
Issue 1: Incoming SIP calls are cut by the OXE after 32 seconds:
- Symptom: Incoming SIP calls are cut by the OXE after ~3 seconds (or 32 seconds in case of
SIP extension) and the 200ok from OXE is never ACK by the external SIP equipment.
- Explanation: If the system is in spatial redundancy, check if the FQDN of the OXE is used by
the external SIP equipement. In fact on the “Contact”, the FQDN is added by the OXE. This
FQDN is unknown by the SIP equipment (because it uses the IP address), and it doesn’t
answer to this 200ok. The OXE sends several times the 200ok and cuts the call because no
ACK is received for this call.
- Solution: The remote SIP equipment must use the FQDN of the OXE. Since the R10, a
parameter is present on the external SIP gateway only “Contact with IP address” used to put
the IP address of the main CPU instead of the FQDN in the Contact header.
- Symptom: The SIP calls are not possible thru an external SIP gateway in high traffic.
- Explanation: Check if the IP address managed on the external SIP gateway is put in
quarantine (in sipalarm files)
- Solution: Manage the IP address on the trusted SIP IP addresses. A restart of the
SIPMOTOR is mandatory after management.
- Symptom: A SIP call, using an ABCF SIP Trunk Group, to an external number is not
possible (thru a carrier for instance) and rejected most of the time by a 502 Bad Gateway.
Internal calls are ok and incoming calls also ok for this SIP equipment.
- Explanation: When the message 502 is reponded to a SIP request, the problem is due to the
management, that means, the information on the SIP request are not good for the call in
progress. In that case, the call is done from an ABCF SIP Trunk Group to an external called
party, the call is rejected because the DID transcoding is set to “True” on the ABCF SIP
Trunk Group
- Solution: Set the “DID transcoding” of the SIP ABCF Trunk group to false (mandatory).
Issue 4: SIP calls are rejected with a 488 Not Acceptable here:
- Explanation: When a SIP call arrives on the OXE, the Call Handling checks if the SDP
received is compatible for this call, if it is not the case, the Call Handling asks the
SIPMOTOR to send a response 488 for this request
- Solution: Manage the SDP of the SIP equipment to be compatible with the configuration of
the OXE or the opposite.
- Explanation: When a SIP call arrives on the OXE, this call is automatically rejected by OXE,
but the reason can be different, even if the scenario of the call is the same. The SIP is linked
to the shelf 19 associated to the CPUs, so if the CPUs are not belonging to the IP domain 0,
the virtual INTIP boards of the shelf 19 doesn’t belong to the IP domain 0, and the SIP is
affected by this configuration.
- Solution: Don't manage CPUs IP addresses on any domain ; they will connect automatically
on the IP domain 0.
- Explanation: When a SIP call is done, a license is used for this call. In case of incoming call,
if no more license is available, the OXE rejects the call by a 403 No licenses available. The
problem can be only the number bought by the customer. It is no enough according to the
number of simultaneous SIP calls, or some SIP call contexts are blocked on the
SIPMOTOR.
- Action to do:
- Solution: If the number of licenses is the number of the licenses bought on OXE, there is no
issue, the solution is to buy more licenses. If the number is less than the number bought,
open a SR and provide the traces files and the Infocollect of the site.
An important thing to remember about SIP device is that all the calls are linked to the SIP Trunk Group
associated to the local SIP Gateway. So if you manage a SIP ABCF Trunk Group or an ISDN SIP Trunk
Group, the behaviour will be different.
Issue 1: Forward on no reply doesn’t work when the destination is a SIP device:
- Symptom: It is not possible to make a forward on no reply (on an IPtouch for instance) when
the destination is a SIP device, ok for immediat forward.
- Explanation: The SIP device behavior is linked to the SIP Trunk group associated to the
local SIP gateway, if you use an ISDN SIP TG, or an ABCF SIP TG, the behaviour will be
different. The SIP Trunk Group used on the local SIP gateway is a SIP ISDN TG.
- Solution: Change the SIP Trung Group managed on the local SIP gateway from SIP ISDN
TG to SIP ABCF TG. A restart of the SIPMOTOR is mandatory.
Issue 2: Afer a while, all SIP phones registrations and subscriptions are impossible
- Symptom: More than 1000 SIP Devices loose their registration. Only a double bascul of
PBX resolves this issue
- Explanation: As there are more than 1000 SIP devices which register/subscribe at the same
time, there is too much traffic to be managed by the PBX and resources on SIPMOTOR are
blocked. Around 45000 Subscription and Registration can be handled in 3 hours time. This
is really a big number. Oxe is dealing with. Solution should be to stop some of the unwanted
Subscribe messages, and increase the subscriptions and registration timers on SIP Devices.
Unwanted subscriptions meant here was even though voice mail was not configured for a
phone set, subscription value was configured, this should be 0.
1348980789 -> Sun Sep 30 06:53:09 2012 SEND MESSAGE TO NETWORK (172.30.125.75:5060 [UDP]) (BUFF LEN =
394)
----------------------utf8-----------------------
SIP/2.0 423 Registration Too Brief
Min-Expires: 1800
- Solutions:
With the current Linux OS, OXE has a limitation in handling more than 1000 data
equipment if it is connected in the same sub-network. So we need to have a seperate
VLAN in between to handle this. OXE CS must be placed under separate subnet and the
IP Phones distributed under different other subnets
The SIP extension is not linked to a SIP Trunk Group, it can be created without SIP management
- Symptom: when a SIP fax equipment tries to make a call, the REINVITE for the T38
negociation is never seen
- Explanation: When a SIP fax call is done, the establishement of the call is done in two
phases, opening of RTP channel then opening of a T38 channel, in case of SIP extension,
the T38 is not implemented, so the second phase cannot be done, and the call is stopped
- Symptom: when a SIP extension is created, it is a multiline user, and if the SIP phone is
associated is monoline, the functioning of the SIP extension can cause issue
- Explanation: A SIP extension user, declared in “business” mode, is multiline, that means taht
teh SIP phone associated must be multiline as well, if it is not the case, the call to the
second line of the user is rejected by the SIP phone, and this can cause disturbances on the
SIP extension behaviour (call handling side) .
Issue 1: One way calls after remote SIP equipment put on hold and call is retrieved:
- Symptom: A SIP call is done between the OXE and a remote SIP gateway. This SIP
equipment puts the call on hold, the OXE equipment can hear the MOH, and when the SIP
equipment retrieves it, the one way call is present.
- Explanation: When the SIP external gateway puts on hold, it sends a REINVITE with a
“Black Hole” (c=0.0.0.0 on SDP) or an “INACTIVE” to stop the RTP flow, before sending a
new REINVITE with a SDP for MOH. When a new REINVITE is sent to get back the
converstaion, the OXE is not able to connect the RTP flow to the SDP given on this
REINVITE.
- Solution: On the external SIP gateway, set the parameter “Ignore inactive/black hole” to
TRUE. In that case, the OXE will not take into account the “Black Hole” or the “INACTIVE”.
- Symptom: An incoming or an outgoing calls are well established, but no speech sent by
OXE
- Explanation: The problem has been seen after an upgrade from a version lower to I160516c
to a R10. On the traces taken, the OXE is not getting SDP or, INVITE or 200ok. The problem
was about the parameter “Routing Application”, this parameter is used for the feature
“Force_on_NET”. In case of incoming call to the OXE, this call is not for an equipment
connected to the OXE, but for an external user (mobile phone for instance), so for such call,
the OXE doesn’t need to reserve ressources on its side. This parameter has been designed
for that.
In case of SIP issue, a minimum of traces must be done, the “motortrace” trace is the minimum. The
Infocollect must always be done in case of SIP issue to get all the information needed to troubleshoot.
If a SR will be opened:
Symptom: SIP ISDN Outgoing call are cancelled by OXE after 180 Ringing SDP (G711) reception.
Solution: As only G711 codec is available for Outgoing calls ( IP Compression Type + G711 on TG) and
caller is located in a restricted domain (Extra Domain Coding Algorithm + With Compression on IP
Domain), OXE cannot sends/receives media stream. Call is cancelled.
Symptom: Re-INVITE sent by OXE to SIP Provider doesn’t contain telephone event media on SDP offer
Solution: On SIP > SIP External Gateway, set parameter “To EMS” to False.
Symptom: Frequent loss of communication between external voicemail and OXE connected via SÏP trunk
Diagnosis: Check if congestion occurs with incident 5816 when you try to access to the voice mail.
Check if Voicemail IP Address is present on Trusted IP Addresses
Solution: Voicemail was put in quarantine and during one half hour all calls in direction of Voicemail were
blocked
13.1.4 Impossible to let a message when routing via SIP Automated Attendant
Symptom: It is not possible to let a message on the voicemail of the called number in case of an automated
attendant SIP and when the Phone Feature COS “Voicemail forwarding” is set at “Ring called set mail”
Solution: On System > Other System Param. > Spec. Customer Features Parameters > Voice Mail
forwarding SIP auto att, set this parameter to true
13.1.5 When call is transfer from a Third Party Server, after few seconds, a Re-Invite is sent by
OXE to reroute RTP to a GD card
Symptom: When call is established, after few seconds, OXE sends a reinvite request to redirect RTP to a GD
card.
Solution: DPNSS is used on this scenario. On System > Other System Param. > External Signalling
Parameters > DeActivate Path Replacement, set this parameter to true
13.1.6 Incoming call from a SIP Third Party Server is rejected by OXE with a SIP Error 488 Not
Acceptable Here
Symptom: Incoming call is rejected by a SIP Error 488 Not acceptable Here
Solution:
On IP > IP Domain > Extra Domain Coding Algorithm must be the same as third party offer
On Categories > Access Category > Go down hierarchy > Public Access Category > Select COS 31
and give correct rights
Symptom: Incoming call received on set phone indicates local call instead of international call.
Diagnosis: - Country code is not separated of received number by PBX so canonical form is not correctly
set up. Canonical form is “+” country code “–” *(number). So, number should be +49–71182137777 in order
to detect that is an international incoming call.
Solution: Add the country code 49 on External Country Code section Translator > External Numbering Plan >
Country Codes:
Country code prefix : 49
Country Value + Germany
13.1.8 When we attempt to register on SIP External Gateway, OXE answers by a SIP error “482
Loop Detected”
Symptom: For each register sent to OXE, we have a SIP error “482 Loop Detected”, as below REGISTER
request:
1352974529 -> Thu Nov 15 11:15:28 2012 SEND MESSAGE TO NETWORK (172.27.139.90:5060 [UDP]) (BUFF LEN
= 478)
----------------------utf8-----------------------
REGISTER sip:hq2cs.labjtr.fr SIP/2.0
Supported: 100rel,path
User-Agent: OmniPCX Enterprise R10.1 j2.501.16.c
To: sip:4321@hq2cs.labjtr.fr
From: sip:4321@hq2cs.labjtr.fr;tag=a9ca34e0b0534fb9d4e0823b7b5d4eaa
Contact: <sip:4321@172.27.145.122;transport=UDP>;expires=1800
Diagnosis: Registration is done by Domain Name resolution so the sip Request-URI sip:hq2cs.labjtr.fr must
be matched with machin name filled on SIP Gateway. The SIP URL of REGISTER contains the SRV/A
domain name. Proxy loops that call back to itself because it does not know about itself as the SRV/A domain.
Solution: Modify the SIP Gateway in order to have the same Machin Name as SIP URL contained on
REGISTER, use the command netadmin to do it:
Trunk Group : 35
IP Address : 172.27.139.90
Machin name : hq2cs.labjtr.fr
Proxy Port Number : 5060
DNS local domain name : labjtr.fr
DNS type + DNS A
First DNS IP Address : 172.27.139.88
13.1.9 When we attempt to register our SIP External Gateway with an external SIP Proxy, SIP
Proxy answers by a SIP error “416 Unsupported URI Scheme”
Symptom: For each register sent to external SIP Proxy, we have a SIP error “416 Unsupported URI
Scheme”, as below REGISTER request:
1352975879 -> Thu Nov 15 11:37:56 2012 RECEIVE MESSAGE FROM NETWORK (172.27.145.122:5060 [UDP])
----------------------utf8-----------------------
REGISTER sip:hq2.labjtr.fr SIP/2.0
Supported: 100rel,path
User-Agent: OmniPCX Enterprise R10.1 j2.501.16.c
To: sip:hq2.labjtr.fr
From: sip:hq2.labjtr.fr;tag=56b8ce5bd76524902b5c171f39c9bbdf
Contact: <sip:172.27.145.122;transport=UDP>;expires=1800
Call-ID: 01f55be7e5c59d21f72659fabc36878a@172.27.145.122
CSeq: 1643105352 REGISTER
Via: SIP/2.0/UDP 172.27.145.122;branch=z9hG4bKdc224f76827da20ba9390b081ef8aed0
Max-Forwards: 70
Content-Length: 0
Diagnosis: Registration ID is not present on REGISTER request so SIP Proxy cannot authenticate the OXE.
Configure the parameter Registration Id on SIP External Gateway
Thu Nov 15 11:45:50 2012 SEND MESSAGE TO NETWORK (172.27.145.122:5060 [UDP]) (BUFF LEN = 396)
----------------------utf8-----------------------
SIP/2.0 200 OK
Contact: <sip:4321@172.27.145.122;transport=UDP>;expires=1800
To: sip:4321@hq2.labjtr.fr;tag=2810b4ed27aa41ba89b99ef3631a8c0d
From: sip:4321@hq2.labjtr.fr;tag=bfc35e619db3ff4f042097e7b390c30a
Call-ID: 5a4750d9baf3b90dd125dccb899bf474@172.27.145.122
CSeq: 571892426 REGISTER
Via: SIP/2.0/UDP 172.27.145.122;branch=z9hG4bK8d42eea8f1c72df626c86ea191f7ff76
Content-Length: 0
13.1.10 Incoming call doesn’t transit via Trunk Group configured on SIP Ext Gw
Symptom: When we make a trkvisu of SIP Trunk Group used by SIP External Gateway during an incoming
call, we observed that no SIP Access is used.
Diagnosis: - by checking INVITE request received from Network, we can see that domain contained on
FROM header is not recognized by SIP External Gateway, so call transits through Main SIP Gateway.
1332292333 -> Wed Mar 21 02:12:13 2012 RECEIVE MESSAGE FROM NETWORK (172.27.138.36:5060 [UDP])
----------------------utf8-----------------------
INVITE sip:11001@172.27.144.20 SIP/2.0
Via: SIP/2.0/UDP 172.27.138.36:5060;branch=z9hG4bK15ac35dc;rport
Max-Forwards: 70
From: "Boss Hoggs" <sip:0033XXXXXXXXX@172.27.144.20>;tag=as5ff02451
To: <sip:11001@172.27.144.20>
Solution: Modify FROM header sent by external application in order to match with remote domain configured
on SIP External Gateway
Symptom: Wrong caller number on OpenTouch anymobile device when using multi device feature.
Example: External user 0980406562 (phone A)
OT MIC SIP directory number 7905 (358306667908) (phone B)
OT anymobile number +358 (0) 505307949 (phone C)
Phone A calls phone B with a redirection to phone C. During phone C ringing phase, Calling Number
of phone B is displayed instead of Calling number of phone A
Diagnosis: - Check if history-info/diversion header is present on requests received from OpenTouch with
related forward informations
- Check External Signalling Parameters (Calling Name Presentation, NPD for external forward
Solution: NPD for external forward is configured at -1 so OXE sends redirecting number in case of forward.
When parameters is configured with NPD used by SIP Trunk Group, initial Calling Number is sent.
Symptom: User A (+33298285305) calls user B (1481001) located on PBX. User B is on immediate forward
to User C (+33675445566). Second leg transits via the Trunk Group 16 (SIP ISDN All Countries) and SIP
External Gw 2 (Remote domain: 172.44.266.44). Diversion header is not added by OXE.
Diagnosis: - Check External Signalling Parameters, Trunk Group and SIP External Gateway configuration
(013064:000323) | Diversion :
(013064:000324) | Url : <> +332675445566@6.1.48.1
(013064:000325) | Reason : UNCONDITIONAL
13.1.13 SIP-Trunking Name is displayed on calling phone set when call is established
Symptom: SIP Trunking Name is displayed on calling phone set when call is established with an external
user through SIP Externl Gateway. SIP Trunk type is ISDN ALL COUNTRIES. Example: A is an internal
phone set and dials external number +33014596222, when call is established, phone set doesn’t display
called number
Diagnosis: Check if SIP Carrier sends a P-Asserted-Identity header on SIP 200 OK Response when call is
established.
Solution: If no Called information is present on connection message (SIP 200 OK), OXE by default displays
the trunk group name.
Diagnosis: When value on From header is not canonical, OXE tags the calling number like ISDN unknown
Solution: Modify the from received on OXE by adding canonical form and manage the country code like this
the calling number will be tagged as national
Diagnosis: As PBX is configured in spatial redundancy, FQDN is used. In this case, FQDN corresponds to
the nodename concatenate with the DNS local domain name managed on SIP Gw. When OXE makes a fax
call to Fax Server, FQDN is used on CONTACT header and as Fax Server cannot resolve it, call is cut.
Solution: Use an external DNS server for FQDN resolution or check at false the “Contact with IP Address”
parameter on SIP Ext Gw.
Diagnosis: On INVITE sent by the FAX Gw, FROM header contains the IP Address of PBX instead of IP
Address of FAX Gw. So, when a Fax call arrives, this is the internal Sip Gw on PBX that is used and SIP-
ABCF trunk group associated. RE-INVITE(T38) is only available on trunk group SIP ISDN.
13.1.17 External call with secret identity over SIP Provider fails
Symptom: Impossible to receive incoming calls with the secret ID
Diagnosis: When a call is received with the secret ID, the call is rejected by OXE with a 480 (not able to
reach the third party)
Solution: The OXE is using the FROM field for the SIP gateway selection, in case of secret id, the FROM
field contains this: anonymous@anonymous.invalid, so an external SIP gateway should correspond to the
domain part of the URI, in that case anonymous.invalid (SIP Remote domain), this external SIP gateway has
the same configuration than the one used to reach the SIP provider.
13.1.18 On SIP outgoing call, dynamic ports are used instead of port 5060
Symptom: why the OXE uses one of the dynamic ports for a SIP call instead of the port 5060?
Diagnosis: When a SIP trace is done with “wireshark”, the source port, when the OXE is the initiator of the
call, can be different from 5060 (SIP port managed on the database)
Solution: Regarding the RFC3581, the initiator of the SIP call can choose a port number different from the
default “SIP port” (5060) for its source port. So in that case the OXE is able to choose one port from the
range of dynamic ports.
The important impacts about this behavior is the management of the size of dynamic ports range and also to
take into accounts the configuration of the firewalls from the customer‘s network, to authorize them to use the
dynamic ports for SIP communication.
13.1.19 A "+" character is added on calling number when ISDN call is routed to SIP
Diagnosis: Addition of "+" is normal, because incoming call from ISDN is tagged with 21 81 which
corresponds to a National Call and according to the RFC, a “+” must be added before the Calling Number
______________________________________________________________________________
| (033539:000002) Concatenated-Physical-Event :
| long: 40 desti: 0 source: 0 cryst: 1 cpl: 6 us: 0 term: 0 type a5
| tei: 0 >>>> message received : SETUP [05] Call ref : 00 37
|______________________________________________________________________________
|
| IE:[04] BEARER_CAPABILITY (l=3) 80 90 a3
| IE:[18] CHANNEL (l=3) a9 83 8c -> T2 : B channel 12 exclusive
| IE:[6c] CALLING_NUMBER (l=6) -> 21 81 Num : 2000
| IE:[7d] HLC (l=2) 91 81
|______________________________________________________________________________
Solution: The "+" is added because the calling party is tagged "national" on the ISDN call, so the OXE ia
added the "+". None configuration must be done on OXE side.
Symptom: User A (+33298285305) calls user B (1481001) located on PBX. User B is on immediate forward
to User C (+33675445566). Second leg transits via the Trunk Group 16 (SIP ISDN All Countries) and SIP
External Gw 2 (Remote domain: 172.44.266.44). Diversion field has not the canonical form: 1481001
Diagnosis: Check NPD configuration, Diversion filed should be as follow: +331481001(canonical format)
corresponds to +33 (France Country Code) 1481001 (Forwarded device number)
Solution: Configure a NPD for normal calls and a NPD for forward as below:
13.1.21 Leg1 and leg2 are external set, when OXE user performs a blind transfer, it doesn’t
work
Symptom: External UserA calls OXE user B thru public SIP Trunk(OXE user DDI: 210457060).
User B calls C (mobile phone) through public SIP trunk
B transfers the call to A before C answers
C answers the call but is not able to talk to external user, transfer is not complete by OXE
Diagnosis: Parameter “Support Re-Invite without SDP” is checked at TRUE on SIP External Gateway.
Consequence is OXE doesn’t perform transfer due to a R&D restriction on support of PRACK by remote
according to this OXE configuration.
Solution: When PRACK is supported by SIP Provider, the parameter “Support Re-Invite without SDP” must
be checked at false on SIP External Gateway.
Solution: Since 10.1 (J2.501.21 release), a new parameter is available on System > Other System Param >
SIP Parameters > Transfer : Refer using single step. This paramter is set by default at True and to obtain
Referred-by in such case, it must be checked at False.
Diagnosis: The following issue is not a problem and is a generic restriction. When SDP received by OXE
exceeds the limit of 1000, INVITE is not duplicate on CPU standby. This allows to avoid problems on
duplication link.
Solution: Change on external application the SDP offer to get only the codec available on the OXE
13.1.24 SIP-Trunking Bad routing and bad display from time to time trough SIP trunk
Symptom: Customer complains of a bad routing of incoming calls from time to time. Also getting strange info
on screen as for example : customer receives " Unavailable " that is displayed on agent desktop and calls
are routed to bad RSI and Agent Group
Diagnosis: SIPMOTOR receives a call with following FROM header: unavailable@unknown.invalid and TO
header 3256391522. As the FROM is wrong formatted, SIPMOTOR cannot find the SIP External Gateway
associated and the SIP Trunk Group.
Nevertheless, the INVITE transits via the Main Gateway (SIP > SIP Gateway) corresponds to virtual entity
1000 on Call Handling:
032042:033267) +------------------------------------------------------------+
(032042:033268) | Message received SIP ----> UA (neqt : 1707)
(032042:033269) | INVITE : +3256391522@10.229.95.250:5060 ; user=phone
(032042:033270) | From : <> unavailable@unknown.invalid:5060 ; user=phone
(032042:033271) | To : <"3256391522 3256391522"> +3256391522@ims.digacom.be:5060 ; user=phone
(032042:033272) +------------------------------------------------------------+
(032042:033273) | SDP :
(032042:033274) | @IP:port = 81.247.255.128:14670
(032042:033275) | ALGOS :
(032042:033276) | PCMA
(032042:033277) | G729
(032042:033278) | DTMF : 101
(032042:033279) | DIRECTION : SEND & RECEIVE
(032042:033280) | cac : false
(032042:033281) | Prack_Required: 0
(032042:033282) | Allow_UPDATE: 0
(032042:033283) | autoAnswer : false
(032042:033284) +------------------------------------------------------------+
(032042:033319) SIP_remp_callin...
When incoming call doesn't match with a SIP External Gateway, default behavior is to send the call on Main
SIP Gateway, Trunk Group used is 59 where no DDI translation is activated so Call Handling take the Called
Number and find on the numbering plan the prefix 3 which corresponds to 2963.. and make the following
SETUP:
CALLING_NUMBER:
CALLED_NUMBER: 296322 => RSI monitored by Call Center
So call is routed to RSI 296322 and calling number cannot be displayed on agent desktop
Solution: Request SIP Provider to resolve the wrong FROM header unavailable@unknown.invalid
The installation consists of 20 external gateways. During the issue, no incidents or backtraces detected but
only incident 5816 Minor failure in SIP component. No “major failure” incidents to report.
Wed Jan 9 06:27:53 2013 11d1-----------------------------------------------------------------
Wed Jan 9 06:27:53 2013 11d1[CMotorCall::onTimersFires] Call (eqt=-1 diag=-1) timer fired type 5.
1357709273 -> Wed Jan 9 06:27:53 2013 11d1---------------------------------------------------------
--------
Wed Jan 9 06:27:53 2013 11d1[CMotorCall::onErrorOnSendRequest] stack::SRM_REGISTER
Wed Jan 9 06:27:53 2013 ALARM: [registerError] failed to emit a Register message.
Diagnosis: We see on provided traces that the ip address 182.16.101.2 is quarantined continuously (4 times
in 2 hrs).
Hence the REGISTER message sent that ip addr. is failed and too many alarms triggerred. Thatswhy motor
goes to degraded mode. This is the main reason for the degraded mode. I checked the infocollect as well as
i loaded the customer database and found that there is no entry in trusted ip:
+-----------------------------------------------------------------------+
| Quaranted IP Address List |
+-----------------------------------------------------------------------+
If we include the ip addresses managed in external gateway in trusted ip then those ips will not be
quarantined. and no REGISTER message will be blocked.
Once you do this, there won’t be much of alarm triggerred and Motor won't go to degraded mode.
Solution: Manage on Trusted IP Addresses all Remote Domain and SIP Outbound Proxies’s IP addresses
used on SIP External Gateway
13.1.26 A Diversion header is added in case of single step transfer after a consultation call
Symptom:
OXE linked to SBC Acme via SIP TG ISDN
OXE linked to SIP Server via SIP TG ABC-F
1) Incoming call through SIP Trunking (ISDN) to a RSI point, strategy route the call to an Agent1.
2) Agent1 makes a consultation call (two step transfer) to the initial RSI point and is in communication with
Agent2.
3) Agent1 or Agent2 releases the call and Agent1 is reconnected to external caller.
4) Agent1 makes a singlesteptransfer to a RSI point which distributes the call to a RoutingPoint monitored by
an external SIP Server.
5) An INVITE is generated by SIPMOTOR to SIPServer and contains an unnecessary history-info header
which contains the RSI used when consultation call.
Diagnosis: According to RFC 5806 Diversion Indication in SIP, this extension provides the ability for the
called SIP user agent to identify from whom the call was diverted and why the call was diverted.
When a diversion occurs, a Diversion header SHOULD be added to the forwarded request or forwarded 3xx
response. The Diversion header MUST contain the Request-URI of the request prior to the diversion.
The Diversion header SHOULD contain a reason that the diversion occurred.
When CSTA function “Diverted” is called by Call Handling, RSI is routing the call to External Routing Point.
Its a kind of diversion (as following figure). Hence, SETUP message will contain
RO_DIVERTING_LEG_INFORMATION2, which will add Diversion Header in Invite.
Solution: Call is diverted by the RSI to an External Routing Point so generated INVITE contains diversion
header. Adding Diversion Header in this scenario is a normal behavior
13.1.27 Incoming calls from SIP Provider are rejected by SIPMOTOR after upgrade from
R9.0 to R10.1
Symptom:
Scenario is the following:
An incoming call from a SIP Provider is handled by OXE Sipmotor and rejected with an error 488 Not
Acceptable Here
Tue Mar 12 09:49:49 2013 RECEIVE MESSAGE FROM NETWORK (194.179.10.3:5060 [UDP])
----------------------utf8-----------------------
INVITE sip:xxxx63324@10.81.32.xxx;user=phone SIP/2.0
Via: SIP/2.0/UDP 194.xxx.10.3:5060;branch=z9hG4bKq5f96100fgbgrtgo72s1.1
To: "xxx163324" <sip:xxx163324@bstk.bifonica.net;user=phone>
From: "Bella Ciao"
<sip:+34xxx163301@bstk.bifonica.net;user=phone>;tag=a1649ecd827305b375fa94a302192f35
Call-ID: ERICSSONBTK_ORIG_10212e20b3bba6afbb51f46cb4bf9515@192.168.195.252
CSeq: 1748174814 INVITE
Max-Forwards: 28
Content-Length: 392
Contact: <sip:+349xxx63301@194.xxx.10.3:5060;transport=udp>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, PRACK, OPTIONS
Supported: timer, 100rel
P-Asserted-Identity: "Bella Ciao" <sip:+349xxx63301@bstk.bifonica.net;user=phone>
User-Agent: OmniPCX Enterprise R10.1
Session-Expires: 600
Min-SE: 180
P-Charging-Vector: icid-value=2257dea5034f1a4d0aa6a336403f0a6;orig-ioi=bifonica.net
Route: <sip:xxx163324@10.81.32.111:5060;user=phone;lr>
Tue Mar 12 09:49:49 2013 114e[CMotorCall::ctrlRouteHeader] call server is in route. ===> the OXE IP
Address is present on Route Header (10.81.32.xxx)
Tue Mar 12 09:49:49 2013 isDomainFromGwExt SCSWorking: NO
Tue Mar 12 09:49:49 2013 [isDomainFromGwExt] Host from request is : bstk.bifonica.net.
Tue Mar 12 09:49:49 2013 [isDomainFromGwExt] User from request is : +34xxx163301
Tue Mar 12 09:49:49 2013 isDomainFromGwExt--> For Non-PCS case GwExt=5
Tue Mar 12 09:49:49 2013 [isValidGwExt] ext gw 5 is valid ===> SIPMOTOR has found the SIP Ext Gw and
Remote Domain matches with the From header [bstk.telefonica.net]
Tue Mar 12 09:49:49 2013 114e[CMotorCall::onReceiveRequest] release the call 488. ==> call is
rejected by SIPMOTOR
Tue Mar 12 09:49:49 2013 SEND MESSAGE TO NETWORK (194.xxx.10.3:5060 [UDP]) (BUFF LEN = 562)
----------------------utf8-----------------------
SIP/2.0 488 Not Acceptable Here
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, SUBSCRIBE, OPTIONS, UPDATE
User-Agent: OmniPCX Enterprise R10.1 j2.501.23
To: "xxx163324" <sip:xxx163324@bstk.bifonica.net;user=phone>;tag=a87ceaccaf57393baca277c6893d0636
From: "Bella Ciao"
<sip:+34xxx163301@bstk.bifonica.net;user=phone>;tag=a1649ecd827305b375fa94a302192f35
Call-ID: ERICSSONBTK_ORIG_10212e20b3bba6afbb51f46cb4bf9515@192.168.195.252
CSeq: 1748174814 INVITE
Diagnosis: Since the release 10.1, a new Boolean has been added on System parameters
Use case 1
INVITE sip:+33155669001@RemoteDomain SIP/2.0
To : <sip:+33155669001@BelongingDomain>
From : <sip:+33147858000@RemoteDomain>
Route : <sip:RegID@OXE_Address> ===> our use case
Although the domain part of the ReqURI doesn’t indicate the OXE, the content of the Route header leads the
OXE to accept the call, thanks to the “loose route” mechanism defined in RFC 3261.
Diagnosis: For call using SIP trunking and other issues, please check that System>Other Parameters :
DTMF on Alert is set to NO.
The default value for "DTMF on Alert" in system parameter is false. For countries, Italy and New Zealand,
this boolean will be set to true defaultly.
13.1.29 Overflow on Remote Extension impossible when SIP Extension seen Out of Service
Symptom: SIP Extension with a Remote Extension tandem (external number thru SIP-Trunking or ISDN)
SIP Extension device is deregistered, out of service on csipsets
When a 4059IP Operatore tries to reach the SIP Extension, overflow to Remote Extension is not happening
13.1.30 Country Code is not added on Calling Number when call is performed since a GSM
Symptom:
On Italy the National Numbering Plan is the following:
- National number: 0xx
- GSM number: 3xx
______________________________________________________________________________
| (958375:000002) Concatenated-Physical-Event :
| long: 54 desti: 0 source: 0 cryst: 4 cpl: 6 us: 0 term: 0 type a5
| tei: 0 >>>> message received : SETUP [05] Call ref : 47 3e
| SENDING COMPLETE
|______________________________________________________________________________
|
| IE:[04] BEARER_CAPABILITY (l=3) 80 90 a3
| IE:[18] CHANNEL (l=3) a9 83 81 -> T2 : B channel 1 exclusive
| IE:[6c] CALLING_NUMBER (l=11) -> 21 80 Num : 117775510 ====> TON National (+39 is added) FROM :
<Isdn_IT> +39117775510@10.64.88.2:5060 ; user=phone
Solution: There is no canonical form in transit when the calling number is Unknown (information received
from Provider for when call is performed from a GSM)
OXE creates a canonical form in transit only with a calling number national or international .
Callin Number Unknown = no modification
Calling Number National = add +xx (xx =country code)
Request provider to send the SETUP with TON National
Issue observed:
User 40x8 or MyIC Desktop makes a Call Back
Invite received by OXE Call Handling is formatted as below:
(076026:000020) | Message received SIP ----> UA (neqt : 2945)
(076026:000021) | INVITE : 0298285305@6.1.48.1:5060 ; user=name
(076026:000022) | From : <HQ148ID4 user> 1481004@otbe.alcatel.ts:5060 ; user=name
(076026:000023) | To : <> 0298285305@otbe.alcatel.ts:5060 ; user=name
First 0 is used by Call Handling for ARS prefix so INVITE generated to provider is formatted like as
298285305 and not routable
We have 00 with first 0 for the ARS Prefix, number sent to SIP Provider is 0298285305
Diagnosis:
Initial INVITE received by OXE is the following:
Tue Mar 19 14:14:53 2013 [display_ipc_out] ------------ Begin ---------------
Tue Mar 19 14:14:53 2013 Id : -1
Tue Mar 19 14:14:53 2013 INVITE
Tue Mar 19 14:14:53 2013 REQUEST URI : <> +33298280001@sip.ale.com:5060 ; user=phone
Tue Mar 19 14:14:53 2013 FROM : <> +33298285305@sip.ale.com:5060 ; user=phone
Tue Mar 19 14:14:53 2013 TO : <"Tango Charlie"> +33298280001@sip.ale.com:5060 ; user=phone
Country Code +33 is received on FROM. Then NPD/External Call Back transforms the number to
00298285305
For Call Back, FROM should be sent to OpenTouch as this: FROM : <0298285305> 00298285305@6.1.48.1:5060 ;
user=phone
13.1.32 only 62 simultaneous calls are sent out of the OXE, all other calls are
released
rd th
Symptom: only 62 simultaneous calls can go out of the OXE, 63 64 ... calls seems to be stuck in the OXE
despite the SIP trunk group shows numerous channels as FREE
Diagnosis: a pair of SIP virtual access is 62 channels. Each time a SIP virtual access is added to a SIP
Trunk group, the Call Server must be rebooted, because these newly created channels will show as FREE
but can’t be used by the Call Handling until a reboot.
Before calling Alcatel’s Business Partner Support Centre (ABPSC), make sure that you have read
through:
The Release Notes which lists features available, restrictions etc.
This chapter and completed the actions suggested for your system’s problem.
Additionally, do the following and document the results so that the Alcatel Technical Support can
better assist you:
Have any information that you gathered while troubleshooting the issue to this point available to
provide to the TAC engineer (such as traces).
[Have a network diagram ready in case of ABC-F Networking problem].
[Have a data network diagram ready in case of VoIP problems. Make sure that relevant information
is listed such as bandwidth of the links, equipments like firewalls, etc.].
[Have a VoIP Audit report available in case of VoIP problems].
Note
Dial-in access is also mandatory to help with effective problem resolution.
Comments
Adapt the paragraph if specific or additional information or actions are required depending on the
subject.
Before the register, make the management of the SIP Gateway & the ABC-F SIP Trunk Group for the
installation of the SIP Processes.
The network used in the SIP TG MUST be different from the one used for the node, the VPN, the TG.
For each SIP Device or SIP Extension, the authentication username and password must be the same in the
OXE management side and SIP set management side
11041 . . . . . OXE
SIP set) (Registrar)
IP=172.27.138.39 FQDN=N11.alcatel.com
| |
|(1) REGISTER |
|-------------------->|
|(2) 401 Unauthorized |
|<--------------------|
|(3) REGISTER |
|-------------------->|
|(4) 200 OK |
|<--------------------|
Challenge explanations :
o The Authentification scheme field corresponds to the OXE information about authentication.
The information “Digest” corresponds to the challenge type
o The information “qop” corresponds to the "quality of protection" values supported by the server.
The value "auth" indicates authentication.
o The information “nonce” corresponds to control the integrity of the authentication information
received by the SIP equipment
o The information “realm” corresponds to the SIP authentication domain, only one can be
managed on the OXE => managed in proxy
The realm is managed in the SIP proxy section, parameter is Authentication realm
| | |
| INVITE | |
|-------------------->| |
| 100 Trying | |
|<--------------------| |
| | Process to contact the callee |
| |<------------------------------->|
| 180 Ringing | |
|<--------------------| |
| 200 OK | |
|<--------------------| |
| ACK | |
|-------------------->| |
| Media Session |
|<=====================================================>|
| BYE | |
|-------------------->| |
| 200 OK | |
|<--------------------| |
| | |
| INVITE | |
|-------------------->| |
| 100 Trying | |
|<--------------------| |
|407 Proxy Auth Required| |
|<--------------------| |
| ACK | |
|-------------------->| |
| | |
|INVITE with challenge| |
|-------------------->| |
| 100 Trying | |
|<--------------------| |
| | Process to contact the callee |
| |<------------------------------->|
| 180 Ringing | |
|<--------------------| |
| 200 OK | |
|<--------------------| |
| ACK | |
|-------------------->| |
| Media Session |
|<=====================================================>|
| BYE | |
|-------------------->| |
| 200 OK | |
|<--------------------| |
Remarks :
The management of the SIP routing on OXE node with ARS & Numbering command table is a
prerequisite and is not included in this documentation
The network used in the ISDN SIP TG MUST be different than the network used for the installation,
the VPN, the ABC SIP TG, the TG.
If a FQDN is used for OXE, you have to do a new netadmin to update correctly the SIP Gateway.
One registration Id is mandatory and the registration timer must be different than 0.
Same scenario with the use of FQDN. As below when FQDN is used for outgoing:
When FQDN is used for incoming, belonging domain parameter must be configured, ex: n12.alcatel.com
In order to REGISTER the external gateway, we need an authentication password managed on OXE.
For that, a creation of a SIP device/SIP Extension user with authentication password is requested. This
step will add the URL and associated password on SIP Dictionnary/SIP Authentication tables used when
a register with challenge is received by sipmotor
Configure the SIP gateway as previously and Configure the SIP proxy :
Authentication
password
We can retrieve the authentication password of this user under : / SIP / Authentication :
When the REGISTER is done, we can see in OXE in user / IP SIP Extension
IP @ of remote
domain
In order to REGISTER the external gateway with the use of a realm, this is exacly the same princip.
We need an authentication password in OXE. For that, a creation of a SIP device user with
authentication password is requested.
Configure the SIP gateway as previously and Configure the SIP proxy :
UAC UAS
12004 N12 N11
11006
(caller). . . . . . .. . . . . . . . . (proxy). . . . . . . . . . .
.(callee)
IP=172.27.144.26 IP=172.27.144.20
Following is the management of authentication on incoming/outgoing calls between two OXE nodes
with the use of FQDN
- END OF DOCUMENT -
156