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OXE6.1:DHCP&3d party SIP clients (sorry if posted twice)

https://alcatelunleashed.com/viewtopic.php?t=932

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OXE6.1:DHCP&3d party SIP clients (sorry if posted twice)

Posted: 06 Aug 2006 11:00
by niko
Hi

I indend to deploy these type of phones in the network:

1. Alcatel IPTouch 4038

2. GrandStream budget tone

The default scope on this OXE is 192.168.1.0/24 and all Alcatel phones are getting the IP Address from OXE DHCP server and correctly hook up to the switch (make incoming, outgoing call ecetera).

I have also created a second scope on the DHCP server, 192.168.15/24 and also have disabled this DHCP setting in OXE mgr: "Only Alcatel Terminals".

However the SIP phones are not receiving an IP Address from the DHCP server, although I see discovery request but no DHCP offer coming out of the OXE switch. I don't see any helpful logging by using motortrace either.

With regard to SIP configuration, this is what I have done so far:

1. Translator / Networking Routing Table -> here I have picked protocol type to QSIG-GF

2. Created a trunk type T2 and set the remote network to 15; Q931 signal to ABC-F; T2 specialty set to SIP; IP Compression set to G 711

3. Set numer of SIP access under "Virtual Access for SIP" to 2

4. In the SIP menu I left default values for both GW and Proxy. Gateway is set to the main IP for the CPU, in my case is 192.168.1.12

5. I created a SIP user, external type and disabled the authentication.

Am I missing or done something wrong so far? What should I look for to get this working.

Thx and have a nice weekend (for whatever is left from it). Niko

Posted: 06 Aug 2006 12:57
by cavagnaro
Your gateway should be you GD IP, not the CS.
mgr/Shelf/Board/Ip Parameters/ check Board 1-0

Posted: 07 Aug 2006 02:05
by alex
Hi folks
First thing is to check whether dhcpd is running
Try this

Code: Select all

(1)a4400a> service  --status-all
MAIN
atd (pid 588) is running...
crond (pid 616) is running...
dhcpd (pid 14352) is running...
4th line will tell you whether dhcpd is running.Or you can use good old "ps -edf | grep dhsp" command.
There is one point about DHCP server configuration on OXE. After each modification you should apply your modifications by choosing "Apply modificatiion" in a menu. E. g. in "mgr" it looks like:

Code: Select all

                   ┌─DHCP Configuration─────────────â”

Posted: 07 Aug 2006 14:26
by niko
cavagnaro wrote:Your gateway should be you GD IP, not the CS.
mgr/Shelf/Board/Ip Parameters/ check Board 1-0
Thanks guys.

DHCP server is running and it serves addresses in the 192.168.1.0/24 subnet. The problem is that the grandstream phone is not able to get an IP from the OXE and am not sure why.

Also I tried to browse trhru the mgr but Im not able to find the GD ip address by following the steps suggested from you Cavagnaro.

Could it be that the menu is different? My OXE runs version 6.1

Thx Niko

Posted: 07 Aug 2006 15:06
by cavagnaro
Hi,

If your app is not reaching to your DHCP server check in : netadmin -m/Security/ Option 2 / Option 1 and check the ranges allowed. This must be done as root (su command)

For the GD Ip address:

Mgr
Shelf
(Go down hierarchy)
(Go down hierarchy)
Ethernet Parameters
Consult Modify
All Instances
Shelf Address = 1
Board Address = 0


Hope this helps

Posted: 08 Aug 2006 15:22
by niko
cavagnaro wrote:Hi,

If your app is not reaching to your DHCP server check in : netadmin -m/Security/ Option 2 / Option 1 and check the ranges allowed. This must be done as root (su command)

For the GD Ip address:

Mgr
Shelf
(Go down hierarchy)
(Go down hierarchy)
Ethernet Parameters
Consult Modify
All Instances
Shelf Address = 1
Board Address = 0


Hope this helps
Ok Guys

I configured the grandstream static so I don't have to deal with this DHCP issue for now (minor, will find the fix for it).

Once I set the Grandstream with a static IP address, the sip client is able to register and there is dial tone also. But if I try to call this SIP extension from an Alcatel IPTouch phone, I get "Out of service" instantly.
If I dial from the SIP phone some other extension then I hear nothing.



Below I have attached the log (level 3 debug) takn from the OXE and the log generate from the phone. Probably the most useful line in the log is "503 Service Unavailable" but I don't know what to look for...

OXE LOG:

SIP/2.0 100 Trying
To: <sip:2002@192.168.1.12>
From: "Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9
Call-ID: a9bca53ce50de5bc@192.168.1.191
CSeq: 63944 INVITE
Via: SIP/2.0/tcp 192.168.1.12;branch=z9hG4bK3aae7657ca0dacd16971414559b38ba8a61134948c173f4ad3ca95ec62386f53
Via: SIP/2.0/UDP 192.168.1.191:6060;branch=z9hG4bKd865d505a9b5ef45
Content-Length: 0

SIP/2.0 503 Service Unavailable
User-Agent: ABS GW v5.1.0
To: <sip:2002@192.168.1.12>;tag=82f4f609fcc728477f72f8c6ed5d68b4
From: "Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9
Call-ID: a9bca53ce50de5bc@192.168.1.191
CSeq: 63944 INVITE
Via: SIP/2.0/tcp 192.168.1.12;branch=z9hG4bK3aae7657ca0dacd16971414559b38ba8a61134948c173f4ad3ca95ec62386f53
Via: SIP/2.0/UDP 192.168.1.191:6060;branch=z9hG4bKd865d505a9b5ef45
Content-Length: 0

-------------------------------------------------

1155065076 -> [CIOCom::receiveResponse] CResponse E ref=1
1155065076 -> [CDispatcher::onIncomingResponse]
1155065076 -> 104 [CCallManager::onIncomingResponse]
1155065076 -> 1107 [CCall::receiveResponse] 100
1155065076 -> 1107 [CCall::getDialog] Confirmed Dialog is not found (ID = d4c4ea8a13f3a9a9;)
1155065076 -> 1509 [CDialog::receiveResponse]
1155065076 -> 2109 [CTransCallingState::receiveResponse] Provisional : Transaction changes to Proceeding state
1155065076 -> 2109 [CTransaction::changeState] STATE CHANGED TO PROCEEDING
1155065076 -> 1509 [CDialog::onTransactionState(pTrans = 2109, previousState = Calling, currentState = Proceeding, reason = 1xx response reception]
1155065076 -> 2109 [CTransaction::freeTimerToken] Timer B is freed
1155065076 -> 2109 [CTransProceedingState::enterInState] Start Timer C
1155065076 -> 2109 [CTransaction::startTimer] Timer C is started (delay = 180000 ms)
1155065076 -> E [CResponse::getProxyContext] Compute proxy context
1155065076 -> C [CRequest::getProxyContext] Get proxy context from received context
1155065076 -> Next message in packet (No 1) :
1155065076 -> RECEIVE MESSAGE FROM NETWORK (192.168.1.12:10018 [tcp])
----------------------utf8-----------------------
SIP/2.0 503 Service Unavailable
User-Agent: ABS GW v5.1.0
To: <sip:2002@192.168.1.12>;tag=82f4f609fcc728477f72f8c6ed5d68b4
From: "Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9
Call-ID: a9bca53ce50de5bc@192.168.1.191
CSeq: 63944 INVITE
Via: SIP/2.0/tcp 192.168.1.12;branch=z9hG4bK3aae7657ca0dacd16971414559b38ba8a61134948c173f4ad3ca95ec62386f53
Via: SIP/2.0/UDP 192.168.1.191:6060;branch=z9hG4bKd865d505a9b5ef45
Content-Length: 0

-------------------------------------------------

1155065076 -> [CIOCom::receiveResponse] CResponse F ref=1
1155065076 -> [CDispatcher::onIncomingResponse]
1155065076 -> 104 [CCallManager::onIncomingResponse]
1155065076 -> 1107 [CCall::receiveResponse] 503
1155065076 -> 1107 [CCall::getDialog] Confirmed Dialog is not found (ID = d4c4ea8a13f3a9a9;82f4f609fcc728477f72f8c6ed5d68b4)
1155065076 -> 1509 [CDialog::receiveResponse]
1155065076 -> E [~CResponse] resp(100) a9bca53ce50de5bc@192.168.1.191 63944 INVITE
1155065076 -> 2109 [CTransProceedingState::receiveResponse] Final : Transaction changes to Completed state
1155065076 -> 2109 [CTransaction::changeState] STATE CHANGED TO COMPLETED
1155065076 -> 1509 [CDialog::onTransactionState(pTrans = 2109, previousState = Proceeding, currentState = Completed, reason = Final resp reception]
1155065076 -> 2109 [CTransaction::freeTimerToken] Timer C is freed
1155065076 -> E [CRequest::CRequest] Creation of a request
1155065076 -> E [CMessage::send] sip_sendMessage (192.168.1.12:6060)
1155065076 -> SEND MESSAGE TO NETWORK (192.168.1.12:6060 [tcp]) (BUFF LEN = 386)
----------------------utf8-----------------------
ACK sip:2002@192.168.1.12:6060;transport=tcp SIP/2.0
Call-ID: a9bca53ce50de5bc@192.168.1.191
From: "Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9
To: <sip:2002@192.168.1.12>;tag=82f4f609fcc728477f72f8c6ed5d68b4
Via: SIP/2.0/tcp 192.168.1.12;branch=z9hG4bK3aae7657ca0dacd16971414559b38ba8a61134948c173f4ad3ca95ec62386f53
CSeq: 63944 ACK
Content-Length: 0

-------------------------------------------------
1155065076 -> info : CTransport::createConnectionTcp() 192.168.1.12 6060
1155065076 -> 2109 [CTransCompletedState::enterInState] Final(!UDP) : INVITE Client Transaction changes to Terminated state
1155065076 -> 2109 [CTransaction::changeState] STATE CHANGED TO TERMINATED
1155065076 -> 1509 [CDialog::updateRouteSet]
1155065076 -> F [CResponse::getProxyContext] Compute proxy context
1155065076 -> C [CRequest::getProxyContext] Get proxy context from received context
1155065076 -> 1107 [CProxyCall::forwardResponse] ---*--- LICENSES ---*--- A reserved license is released the INVITE is rejected
1155065076 -> 1107 [CCall::makeGenericResponse] 503
1155065076 -> 1107 [CCall::getDialog] Confirmed Dialog is not found (ID = 82f4f609fcc728477f72f8c6ed5d68b4;d4c4ea8a13f3a9a9)
1155065076 -> 1508 [CDialog::createResponse] 503
1155065076 -> 2108 [CTransProceedingState::createResponse] CResponse 10 ref=1
1155065076 -> 10 [CMessage::send] sip_sendMessage (192.168.1.191:6060)
1155065076 -> SEND MESSAGE TO NETWORK (192.168.1.191:6060 [udp]) (BUFF LEN = 354)
----------------------utf8-----------------------
SIP/2.0 503 Service Unavailable
User-Agent: ABS GW v5.1.0
To: <sip:2002@192.168.1.12>;tag=82f4f609fcc728477f72f8c6ed5d68b4
From: "Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9
Call-ID: a9bca53ce50de5bc@192.168.1.191
CSeq: 63944 INVITE
Via: SIP/2.0/UDP 192.168.1.191:6060;branch=z9hG4bKd865d505a9b5ef45
Content-Length: 0

-------------------------------------------------
1155065076 -> info : CTransport::createConnectionUdp 192.168.1.191 6060
1155065076 -> B [~CResponse] resp(100) a9bca53ce50de5bc@192.168.1.191 63944 INVITE
1155065076 -> 2108 [CTransProceedingState::createResponse] Final : Transaction changes to Completed state
1155065076 -> 2108 [CTransaction::changeState] STATE CHANGED TO COMPLETED
1155065076 -> 1508 [CDialog::onTransactionState(pTrans = 2108, previousState = Proceeding, currentState = Completed, reason = Final resp creation]
1155065076 -> 2108 [CTransaction::startTimer] Timer G is started (delay = 500 ms)
1155065076 -> 2108 [CTransaction::startTimer] Timer H is started (delay = 32000 ms)
1155065076 -> 1508 [CDialog::updateRouteSet]
1155065076 -> info : CTransport::DoSTest(192.168.1.12,10017,1) NbMsg:2 < seuil
1155065076 -> info : CTransport::receiveTcpProc() from 192.168.1.12:10017 prot:1 rc:386 msg:2
1155065076 -> RECEIVE MESSAGE FROM NETWORK (192.168.1.12:10017 [tcp])
----------------------utf8-----------------------
ACK sip:2002@192.168.1.12:6060;transport=tcp SIP/2.0
Call-ID: a9bca53ce50de5bc@192.168.1.191
From: "Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9
To: <sip:2002@192.168.1.12>;tag=82f4f609fcc728477f72f8c6ed5d68b4
Via: SIP/2.0/tcp 192.168.1.12;branch=z9hG4bK3aae7657ca0dacd16971414559b38ba8a61134948c173f4ad3ca95ec62386f53
CSeq: 63944 ACK
Content-Length: 0

-------------------------------------------------

1155065076 -> F [CRequest::CRequest] Creation of a request (RECEIVED FROM NETWORK)
1155065076 -> [CIOCom::receiveRequest] Received CRequest F ref=1
1155065076 -> [CDispatcher::onIncomingRequest()]
1155065076 -> 102 [CCallManager::onIncomingRequest()]
1155065076 -> 1108 [CCall::receiveRequest] ACK
1155065076 -> 1108 [CCall::getDialog] Confirmed Dialog is not found (ID = 82f4f609fcc728477f72f8c6ed5d68b4;d4c4ea8a13f3a9a9)
1155065076 -> 150a [CDialog::receiveRequest]
1155065076 -> 150a [CDialog::receiveAckRequest]
1155065076 -> 210a [CTransCompletedState::receiveRequest] ACK(UDP) : Transaction changes to Confirmed state
1155065076 -> 210a [CTransaction::changeState] STATE CHANGED TO CONFIRMED
1155065076 -> 150a [CDialog::onTransactionState(pTrans = 210a, previousState = Completed, currentState = Confirmed, reason = Request reception]
1155065076 -> 210a [CTransaction::freeTimerToken] Timer H is freed
1155065076 -> 210a [CTransaction::changeState] STATE CHANGED TO TERMINATED
1155065076 -> 150a [CDialog::receiveAckRequest] receiving an ACK on a previously rejected INVITE
1155065076 -> 1108 [CCall::checkAuthentication] ACK
1155065076 -> [CMotorCall] same Dialog
1155065076 -> [onReceiveRequest] Ack => update the inviteContext.
1155065076 -> [receiveAckMessage] Call: a9bca53ce50de5bc@192.168.1.191 eqt: 1331 TERMINATED_STATE received a message.
1155065076 -> [ipc_thread] IPC Thread : Ipc reception.
1155065076 -> [ipc_thread] IPC Thread : TCL thread signaled.
1155065076 -> [emitEventToMonitel] Event sent on eqt : 1331
1155065076 -> [display_ipc_out] ------------ Begin ---------------
1155065076 -> ACK
1155065076 -> [display_ipc_out] ------------- End ----------------
1155065076 -> [receiveAckMessage] Reset of the session.
1155065076 -> [CMotorCall::unRegister] Remove eqt : 1331 from the map.
1155065076 -> [CMotorCallManager::eraseCallwithEqt] CMotorCall 1331 erased.
1155065076 -> [exec_ipc] in.
1155065076 -> [display_ipc_in] ------------ Begin ---------------
1155065076 -> neqt : 1331
1155065076 -> SIP EQT RELEASED
1155065076 -> [display_ipc_in] ------------- End ----------------
1155065076 -> [CMotorCallManager::onIncomingEvent] an event 10773 arrived on the eqt 1331.
1155065076 -> [CMotorCallManager::onIncomingEvent] NO call with eqt: 1331 to release.
1155065076 -> info : CTransport::receiveUdpProc() from 192.168.1.191:44055 prot:0 rc:488 msg:9
1155065076 -> info : CTransport::DoSTest(192.168.1.191,44055,0) NbMsg:9 < seuil
1155065077 -> RECEIVE MESSAGE FROM NETWORK (192.168.1.191:44055 [udp])
----------------------utf8-----------------------
ACK sip:2002@192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:6060;branch=z9hG4bKd865d505a9b5ef45
From: "Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9
To: <sip:2002@192.168.1.12>;tag=82f4f609fcc728477f72f8c6ed5d68b4
Contact: <sip:5000@192.168.1.191:6060>
Call-ID: a9bca53ce50de5bc@192.168.1.191
CSeq: 63944 ACK
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

-------------------------------------------------

1155065077 -> 10 [CRequest::CRequest] Creation of a request (RECEIVED FROM NETWORK)
1155065077 -> [CIOCom::receiveRequest] Received CRequest 10 ref=1
1155065077 -> [CDispatcher::onIncomingRequest()]
1155065077 -> 104 [CCallManager::onIncomingRequest()]
1155065077 -> 1107 [CCall::receiveRequest] ACK
1155065077 -> 1107 [CCall::getDialog] Confirmed Dialog is not found (ID = 82f4f609fcc728477f72f8c6ed5d68b4;d4c4ea8a13f3a9a9)
1155065077 -> 1508 [CDialog::receiveRequest]
1155065077 -> 1508 [CDialog::receiveAckRequest]
1155065077 -> 2108 [CTransCompletedState::receiveRequest] ACK(UDP) : Transaction changes to Confirmed state
1155065077 -> 2108 [CTransaction::changeState] STATE CHANGED TO CONFIRMED
1155065077 -> 1508 [CDialog::onTransactionState(pTrans = 2108, previousState = Completed, currentState = Confirmed, reason = Request reception]
1155065077 -> 2108 [CTransaction::freeTimerToken] Timer G is freed
1155065077 -> 2108 [CTransaction::freeTimerToken] Timer H is freed
1155065077 -> 2108 [CTransaction::startTimer] Timer I is started (delay = 5000 ms)
1155065077 -> 1508 [CDialog::receiveAckRequest] receiving an ACK on a previously rejected INVITE
1155065077 -> 1509 [CDialog::onTransactionState(pTrans = 2109, previousState = null, currentState = Terminated, reason = Timer D fires]
1155065077 -> 1509 [CDialog::onTransactionState] dialog is terminated
1155065077 -> 1509 [CDialog::onTransactionState] onDialogClosed
1155065077 -> 1107 [CCall::onDialogClosed]
1155065077 -> 1107 [CCall::onDialogClosed] only initial *********
1155065077 -> E [~CRequest] req a9bca53ce50de5bc@192.168.1.191 63944 ACK
1155065077 -> F [~CResponse] resp(503) a9bca53ce50de5bc@192.168.1.191 63944 INVITE
1155065077 -> 150a [CDialog::onTransactionState(pTrans = 210a, previousState = null, currentState = Terminated, reason = Timer I fires]
1155065077 -> 150a [CDialog::onTransactionState] dialog is terminated
1155065077 -> 150a [CDialog::onTransactionState] onDialogClosed
1155065077 -> 1108 [CCall::onDialogClosed]
1155065077 -> 1108 [CCall::onDialogClosed] only initial *********
1155065077 -> 1108 [CCall::onDialogClosed] All the initial dialogs in the Call are closed
1155065077 -> 1108 [CCall] closed
1155065077 -> 102 [CCallManager::onCallClosed]
1155065077 -> [CMotorCallManager::onNotifyCallClosed] Call : a9bca53ce50de5bc@192.168.1.191 is closed.
1155065077 -> F [~CRequest] req a9bca53ce50de5bc@192.168.1.191 63944 ACK
1155065078 -> info : CTransport::receiveUdpProc() from 192.168.1.191:44055 prot:0 rc:4 msg:10
1155065078 -> info : CTransport::DoSTest(192.168.1.191,44055,0) NbMsg:10 < seuil
1155065078 -> RECEIVE MESSAGE FROM NETWORK (192.168.1.191:44055 [udp])
----------------------utf8-----------------------
ÀÀÀÀ-------------------------------------------------

1155065078 -> [CIOCom::onReceiveMessage] Message is incomplete
1155065078 -> [CIOCom::getMalformedMessage]
1155065078 -> [CIOCom::getMalformedMessage] sip_decodeMessageWithoutCallback FAILED => HSS ERROR Incomplete
1155065079 -> 1108 [CCall::removeGarbage]
1155065079 -> 150a [CDialog::~CDialog]
1155065079 -> [---*--- SESSION-TIMER ---*---] Free CSessionTimerContext

1155065079 -> [CMotorCall::~CMotorCall] Call : a9bca53ce50de5bc@192.168.1.191 is being destructed.
1155065079 -> 1108 [CCall::~CCall]
1155065079 -> 1108 [CCall::removeGarbage]
1155065079 -> D [~CResponse] resp(503) a9bca53ce50de5bc@192.168.1.191 63944 INVITE
1155065079 -> D [~CRequest] req a9bca53ce50de5bc@192.168.1.191 63944 INVITE
1155065081 -> 2108 [CTransConfirmedState::timerFires] Timer I fires
1155065081 -> 2108 [CTransConfirmedState::timerFires] TimerI : Transaction changes to Terminated state
1155065081 -> 2108 [CTransaction::changeState] STATE CHANGED TO TERMINATED
1155065082 -> 1508 [CDialog::onTransactionState(pTrans = 2108, previousState = null, currentState = Terminated, reason = Timer I fires]
1155065082 -> 1508 [CDialog::onTransactionState] dialog is terminated
1155065082 -> 1508 [CDialog::onTransactionState] onDialogClosed
1155065082 -> 1107 [CCall::onDialogClosed]
1155065082 -> 1107 [CCall::onDialogClosed] only initial *********
1155065082 -> 1107 [CCall::onDialogClosed] All the initial dialogs in the Call are closed
1155065082 -> 1107 [CCall] closed
1155065082 -> 104 [CCallManager::onCallClosed]
1155065082 -> 10 [~CRequest] req a9bca53ce50de5bc@192.168.1.191 63944 ACK

(1)cpu


GRANDSTREAM PHONE:

08-08-2006 15:15:56 User.Debug 192.168.1.191 GS_LOG: [00.0B.82.05.CA.AE][000] ACK sip:2002@192.168.1.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.191:6060;branch=z9hG4bKd865d505a9b5ef45 From: "Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9 To: <sip:2002@192.168.1.12>;tag=82f4f609fcc728477f72f8c6ed5d68b4 Contact: <sip:5000@192.168.1.191:6060> Call-ID: a9bca53ce50de5bc@192.168.1.191 CSeq: 63944 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0
08-08-2006 15:15:56 User.Info 192.168.1.191 GS_LOG: [00.0B.82.05.CA.AE][000] Send SIP message: 3 To 192.168.1.12:5060
08-08-2006 15:15:56 User.Info 192.168.1.191 GS_LOG: [00.0B.82.05.CA.AE][000] Received SIP message: 503
08-08-2006 15:15:56 User.Debug 192.168.1.191 GS_LOG: [00.0B.82.05.CA.AE][000] SIP/2.0 503 Service Unavailable User-Agent: ABS GW v5.1.0 To: <sip:2002@192.168.1.12>;tag=82f4f609fcc728477f72f8c6ed5d68b4 From: "Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9 Call-ID: a9bca53ce50de5bc@192.168.1.191 CSeq: 63944 INVITE Via: SIP/2.0/UDP 192.168.1.191:6060;branch=z9hG4bKd865d505a9b5ef45 Content-Length: 0
08-08-2006 15:15:55 User.Info 192.168.1.191 GS_LOG: [00.0B.82.05.CA.AE][000] Received SIP message: 100
08-08-2006 15:15:55 User.Debug 192.168.1.191 GS_LOG: [00.0B.82.05.CA.AE][000] SIP/2.0 100 Trying To: <sip:2002@192.168.1.12> From: "Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9 Call-ID: a9bca53ce50de5bc@192.168.1.191 CSeq: 63944 INVITE Via: SIP/2.0/UDP 192.168.1.191:6060;branch=z9hG4bKd865d505a9b5ef45 Content-Length: 0
08-08-2006 15:15:55 User.Info 192.168.1.191 GS_LOG: [00.0B.82.05.CA.AE][000] INVITE From="Niko - SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9 To=<sip:2002@192.168.1.12>
08-08-2006 15:15:55 User.Debug 192.168.1.191 GS_LOG: [00.0B.82.05.CA.AE][000] INVITE sip:2002@192.168.1.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.191:6060;branch=z9hG4bKd865d505a9b5ef45 From: "SIP Phone" <sip:5000@192.168.1.12>;tag=d4c4ea8a13f3a9a9 To: <sip:2002@192.168.1.12> Contact: <sip:5000@192.168.1.191:6060> Supported: replaces Call-ID: a9bca53ce50de5bc@192.168.1.191 CSeq: 63944 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 410 v=0 o=5000 8000 8000 IN IP4 192.168.1.191 s=SIP Call c=IN IP4 192.168.1.191 t=0 0 m=audio 8004 RTP/AVP 0 8 4 18 2 15 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11

Posted: 08 Aug 2006 17:18
by cavagnaro
i have the same problem but with a hardware sip phone. I still can figure out why, the problem i see is that the software is not too complete as other sip phones like Xlite. Maybe you are having the same problem on client side.

Posted: 09 Aug 2006 09:40
by niko
cavagnaro wrote:i have the same problem but with a hardware sip phone. I still can figure out why, the problem i see is that the software is not too complete as other sip phones like Xlite. Maybe you are having the same problem on client side.
My SIP phone is also hardware based, grand stream VT 100.

Posted: 09 Aug 2006 12:34
by cavagnaro
Mine is a Vizufon. Humm, i tested this:

A hardware Sip Phone and a Software Sip Phone.

Software sip can call digital set.
Software sip can call hardware sip set.

Digital set can call software sip.
Digital set can't call hardware sip.

Hardware sip can call digital set.
Hardware sip can call software sip.

So i'm more confused now. Why between SIPs the call proceed and why doesn't happen with digital sets as well.

Posted: 09 Aug 2006 13:26
by niko
cavagnaro wrote:Mine is a Vizufon. Humm, i tested this:

A hardware Sip Phone and a Software Sip Phone.

Software sip can call digital set.
Software sip can call hardware sip set.

Digital set can call software sip.
Digital set can't call hardware sip.

Hardware sip can call digital set.
Hardware sip can call software sip.

So i'm more confused now. Why between SIPs the call proceed and why doesn't happen with digital sets as well.
I have gone this far for now:

1. I can successfully make or receive calls between two SIP sets (either hardware or software based clients).
2. I can initiate a call from an Alcatel IP phone to a SIP phone, acknowldege the call in the SIP phone, but then the call just disconnects.

I can't, still, call from a SIP phone to a IP Touch phone. I always get the famous messge: "Service unavailable 503".

Any idea what this error is for?

-Niko
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