Trouble making and recieving calls on SIP

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alcatelguru
Member
Posts: 29
Joined: 04 Mar 2008 19:16

Re: Trouble making and recieving calls on SIP

Post by alcatelguru »

Hello,
I have made the compression without and still facing problem.

(1)xa000000> compvisu sys────────────────────────────────────────────────┘

Thu Oct 23 19:26:21 EST 2008

C O M P V I S U |
+==============================================================================+
| Inter-node protocol H323....... yes |
| RTP Direct..................... yes |
| RTP Direct for H323 terminals.. no |
| Fast Start..................... yes |
| VAD (Voice Activity Detection): |
| - G723/G729...... yes |
| - G711........... no |
| ECE (Echo Canceller)........... yes |
| - LIO/LIOE........ 16 ms |
| - INTIP/GA/GD..... 128 ms |
| PFE (Post Filter).............. yes |
| Volume ........................ 8 |
| Volume for IP Phone ........... 0dB |
| Volume for other device. ...... 0dB |
| VRE ........................... no |
| Law (Except Media Gateway)..... A law |
| Global compression type ....... G723 |
| Multi-algorithm (for H323) .... no |
| Compression for INTIP/GD ..... without |
| Compression for IPP ........... without |
| Transit on IP Boards ...........yes |
| ticket Stat IP................. no |
| IP version..................... IPv4 |
| Transit compatibility.......... no |
| Voip Framing G711 ............. 30 ms |
| Voip Framing G723 ............. 30 ms |
| Voip Framing G729 ............. 20 ms |
| No RBT For Direct RTP H323..... no |
+=====================================================================

I have changed the Trunk COS and also GD set.

Still facing problem.

Thanks
alcatelguru
Member
Posts: 29
Joined: 04 Mar 2008 19:16

Re: Trouble making and recieving calls on SIP

Post by alcatelguru »

Latest Compvisu sys

(1)xa000000> compvisu sys

Thu Oct 23 19:36:51 EST 2008

+==============================================================================+
| C O M P V I S U |
+==============================================================================+
| Inter-node protocol H323....... yes |
| RTP Direct..................... yes |
| RTP Direct for H323 terminals.. no |
| Fast Start..................... yes |
| VAD (Voice Activity Detection): |
| - G723/G729...... yes |
| - G711........... yes |
| ECE (Echo Canceller)........... yes |
| - LIO/LIOE........ 16 ms |
| - INTIP/GA/GD..... 128 ms |
| PFE (Post Filter).............. yes |
| Volume ........................ 8 |
| Volume for IP Phone ........... 0dB |
| Volume for other device. ...... 0dB |
| VRE ........................... no |
| Law (Except Media Gateway)..... A law |
| Global compression type ....... G723 |
| Multi-algorithm (for H323) .... no |
| Compression for INTIP/GD ..... without |
| Compression for IPP ........... without |
| Transit on IP Boards ...........yes |
| ticket Stat IP................. no |
| IP version..................... IPv4 |
| Transit compatibility.......... no |
| Voip Framing G711 ............. 30 ms |
| Voip Framing G723 ............. 30 ms |
| Voip Framing G729 ............. 20 ms |
| No RBT For Direct RTP H323..... no |
+==============================================================================+
alcatelguru
Member
Posts: 29
Joined: 04 Mar 2008 19:16

Re: Trouble making and recieving calls on SIP

Post by alcatelguru »

Hello,
If I have configured wrong,Can you login to have a check on my configuration and advise
Ip Address: 203.36.12.10

Thanks for your help.
alcatelboy
Member
Posts: 146
Joined: 27 Dec 2005 06:46
Location: south of china
Contact:

Re: Trouble making and recieving calls on SIP

Post by alcatelboy »

fill the realm ,account and the password in the ext gateway . use the ARS prefix to dial.
AQPS ACFE ACSE
alcatelguru
Member
Posts: 29
Joined: 04 Mar 2008 19:16

Re: Trouble making and recieving calls on SIP

Post by alcatelguru »

Hi,
There is no realm or account or password.There is no authentication.I have used the prefix to dial out.but still facing the problem.

Thanks
alcatelguru
Member
Posts: 29
Joined: 04 Mar 2008 19:16

Re: Trouble making and recieving calls on SIP

Post by alcatelguru »

New Outgoing calls traces,If this helps

Calling extension:3901 Full number:0290083901
Called number:0437407840


(1)xa000000> motortrace 2
motortrace (v5.1.0) verbosity = 0003b004
sipmotor trace-level set 2 (Medium trace).
(1)xa000000> traced
** UNIX-trace-daemon started ... (static user group No 1) **

traced started ...
1224814857 -> DECT: get_redial_nb_cells nb_cells=3 tiny=0
1224814857 -> [exec_ipc] in.
1224814857 -> [display_ipc_in] ------------ Begin ---------------
1224814857 -> neqt : 1036
1224814857 -> INVITE
1224814857 -> REQUEST URI : <> 0437407840@203.36.12.10:5060 ; user=phone
1224814857 -> FROM : <0290083901> +0290083901@203.36.12.10:5060 ; user=name
1224814857 -> TO : <> 0437407840@203.36.12.10:5060 ; user=phone
1224814857 -> CAC : 0
1224814857 -> neqt CAC : -1
1224814857 -> CLIR : 0
1224814857 -> SDP :
1224814857 -> ADDRESS : 203.36.12.2 :32514
1224814857 -> ALGOS :
1224814857 -> PCMA
1224814857 -> PCMU
1224814857 -> G723
1224814857 -> 97
1224814857 -> DIRECTION : SEND & RECEIVE
1224814857 -> N_GW_EXT : 1000
1224814857 -> [display_ipc_in] ------------- End ----------------
1224814857 -> [CMotorCallManager::onIncomingEvent] an event 10752 arrived on the
eqt 1036.
1224814857 -> [CMotorCallManager::onIncomingEvent] New Call created.
1224814857 -> [CMotorCallManager::onIncomingEvent] eqt=1036.
1224814857 -> [CMotorCallManager::onIncomingEvent] state = INITIAL_STATE.
1224814857 -> [onIncomingEvent] INVITE arrived.
1224814857 -> [receiveInviteEvent] Call: eqt: 1036 INITIAL_STATE received an ev
ent.
1224814857 -> 110b[CMotorCall::makeResponseSdp] format 97
1224814857 -> 110b[CMotorCall::new_determineTargets] try with alias.
1224814857 -> [getUrlByAlias] user=0437407840 domain=203.36.12.10.
1224814857 -> [getUrlByAlias] NOK
1224814857 -> 110b[CMotorCall::makeInitialMotorRequest]-->extGw=1000
1224814857 -> SEND MESSAGE TO NETWORK (203.36.12.10:5060 [UDP]) (BUFF LEN = 816)
----------------------utf8-----------------------
INVITE sip:0437407840@203.36.12.10;user=phone SIP/2.0
Supported: replaces,100rel
User-Agent: ABS GW v5.1.0
Content-Type: application/sdp
To: <sip:0437407840@203.36.12.10;user=phone>
From: "0290083901" <sip:+0290083901@203.36.12.10>;tag=759545f456b0f992e60dc0549a
a14bff
Contact: sip:203.36.12.10
Call-ID: 44a44fd5b99faeb43d149013fe1ff30b@203.36.12.10
CSeq: 292214136 INVITE
Via: SIP/2.0/UDP 203.36.12.10;branch=z9hG4bK90f251b0f06434497eb3274cfd5b613a
Max-Forwards: 70
Content-Length: 315

v=0
o=OXE 1224814856 1224814856 IN IP4 203.36.12.10
s=abs
c=IN IP4 203.36.12.2
t=0 0
m=audio 32514 RTP/AVP 8 0 4 97
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:30
a=rtpmap:4 G723/8000
a=ptime:30
a=maxptime:30
a=rtpmap:97 telephone-event/8000
-------------------------------------------------
1224814857 -> [onIncomingEvent] CallID must be inserted.
1224814857 -> [onIncomingEvent] CallID inserted.
1224814857 -> RECEIVE MESSAGE FROM NETWORK (203.36.12.10:4135 [UDP])
----------------------utf8-----------------------
INVITE sip:0437407840@203.36.12.10;user=phone SIP/2.0
Supported: replaces,100rel
User-Agent: ABS GW v5.1.0
Content-Type: application/sdp
To: <sip:0437407840@203.36.12.10;user=phone>
From: "0290083901" <sip:+0290083901@203.36.12.10>;tag=759545f456b0f992e60dc0549a
a14bff
Contact: sip:203.36.12.10
Call-ID: 44a44fd5b99faeb43d149013fe1ff30b@203.36.12.10
CSeq: 292214136 INVITE
Via: SIP/2.0/UDP 203.36.12.10;branch=z9hG4bK90f251b0f06434497eb3274cfd5b613a
Max-Forwards: 70
Content-Length: 315

v=0
o=OXE 1224814856 1224814856 IN IP4 203.36.12.10
s=abs
c=IN IP4 203.36.12.2
t=0 0
m=audio 32514 RTP/AVP 8 0 4 97
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:30
a=rtpmap:4 G723/8000
a=ptime:30
a=maxptime:30
a=rtpmap:97 telephone-event/8000
-------------------------------------------------

1224814857 -> SEND MESSAGE TO NETWORK (203.36.12.10:5060 [UDP]) (BUFF LEN = 333)
----------------------utf8-----------------------
SIP/2.0 100 Trying
To: <sip:0437407840@203.36.12.10;user=phone>
From: "0290083901" <sip:+0290083901@203.36.12.10>;tag=759545f456b0f992e60dc0549a
a14bff
Call-ID: 44a44fd5b99faeb43d149013fe1ff30b@203.36.12.10
CSeq: 292214136 INVITE
Via: SIP/2.0/UDP 203.36.12.10;branch=z9hG4bK90f251b0f06434497eb3274cfd5b613a
Content-Length: 0

-------------------------------------------------
1224814857 -> 110b[CMotorCall::toGatewayOrProxy] request for proxydomain=203.36.
12.10.
1224814857 -> [isUserRegistred] getRegistration=0437407840.
1224814857 -> [isUserRegistred] result=-1.
1224814857 -> 110b[CMotorCall::onReceiveRequest] request for gateway.
1224814857 -> 110b[CMotorCall::onReceiveRequest] state = PROCEEDING_STATE.
1224814857 -> 110b[CMotorCall::methodInviteReceived] CAC = 0
1224814857 -> [receiveInviteMessage] incoming call.
1224814857 -> SEND MESSAGE TO NETWORK (203.36.12.10:5060 [UDP]) (BUFF LEN = 478)
----------------------utf8-----------------------
SIP/2.0 488 Not Acceptable Here
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, SUBSCRIBE, OPTIONS
User-Agent: ABS GW v5.1.0
To: <sip:0437407840@203.36.12.10;user=phone>;tag=491dfd12089985ac4c9f9562fb8ac06
4
From: "0290083901" <sip:+0290083901@203.36.12.10>;tag=759545f456b0f992e60dc0549a
a14bff
Call-ID: 44a44fd5b99faeb43d149013fe1ff30b@203.36.12.10
CSeq: 292214136 INVITE
Via: SIP/2.0/UDP 203.36.12.10;branch=z9hG4bK90f251b0f06434497eb3274cfd5b613a
Content-Length: 0

-------------------------------------------------
1224814857 -> RECEIVE MESSAGE FROM NETWORK (203.36.12.10:4135 [UDP])
----------------------utf8-----------------------
SIP/2.0 100 Trying
To: <sip:0437407840@203.36.12.10;user=phone>
From: "0290083901" <sip:+0290083901@203.36.12.10>;tag=759545f456b0f992e60dc0549a
a14bff
Call-ID: 44a44fd5b99faeb43d149013fe1ff30b@203.36.12.10
CSeq: 292214136 INVITE
Via: SIP/2.0/UDP 203.36.12.10;branch=z9hG4bK90f251b0f06434497eb3274cfd5b613a
Content-Length: 0

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

1224814857 -> [receiveInformationalMessage] Call: 44a44fd5b99faeb43d149013fe1ff3
0b@203.36.12.10 eqt: 1036 TERMINATED_STATE received a not handled message.
1224814857 -> RECEIVE MESSAGE FROM NETWORK (203.36.12.10:4135 [UDP])
----------------------utf8-----------------------
SIP/2.0 488 Not Acceptable Here
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, SUBSCRIBE, OPTIONS
User-Agent: ABS GW v5.1.0
To: <sip:0437407840@203.36.12.10;user=phone>;tag=491dfd12089985ac4c9f9562fb8ac06
4
From: "0290083901" <sip:+0290083901@203.36.12.10>;tag=759545f456b0f992e60dc0549a
a14bff
Call-ID: 44a44fd5b99faeb43d149013fe1ff30b@203.36.12.10
CSeq: 292214136 INVITE
Via: SIP/2.0/UDP 203.36.12.10;branch=z9hG4bK90f251b0f06434497eb3274cfd5b613a
Content-Length: 0

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

1224814857 -> SEND MESSAGE TO NETWORK (203.36.12.10:5060 [UDP]) (BUFF LEN = 399)
----------------------utf8-----------------------
ACK sip:0437407840@203.36.12.10;user=phone SIP/2.0
Call-ID: 44a44fd5b99faeb43d149013fe1ff30b@203.36.12.10
From: "0290083901" <sip:+0290083901@203.36.12.10>;tag=759545f456b0f992e60dc0549a
a14bff
To: <sip:0437407840@203.36.12.10;user=phone>;tag=491dfd12089985ac4c9f9562fb8ac06
4
Via: SIP/2.0/UDP 203.36.12.10;branch=z9hG4bK90f251b0f06434497eb3274cfd5b613a
CSeq: 292214136 ACK
Content-Length: 0

-------------------------------------------------
1224814857 -> [receiveRequestFailureMessage] Call: 44a44fd5b99faeb43d149013fe1ff
30b@203.36.12.10 eqt: 1036 TERMINATED_STATE received a message.
1224814857 -> 110b[CMotorCall::emitEventToMonitel] content size (lginfo): 652
1224814857 -> 110b[CMotorCall::emitEventToMonitel] CAC = 0 eqtCAC = -1
1224814857 -> 110b[CMotorCall::emitEventToMonitel] CAC = 0
1224814857 -> 110b[CMotorCall::emitEventToMonitel] Event sent on eqt : 1036
1224814857 -> [display_ipc_out] ------------ Begin ---------------
1224814857 -> REQUEST FAILURE
1224814857 -> xx : 0
1224814857 -> RELATIVE REQUEST : INVITE
1224814857 -> [display_ipc_out] ------------- End ----------------
1224814857 -> [exec_ipc] in.
1224814857 -> [display_ipc_in] ------------ Begin ---------------
1224814857 -> neqt : 1036
1224814857 -> SIP EQT RELEASED
1224814857 -> [display_ipc_in] ------------- End ----------------
1224814857 -> [CMotorCallManager::onIncomingEvent] an event 10773 arrived on the
eqt 1036.
1224814857 -> [CMotorCallManager::onIncomingEvent] The call with eqt: 1036 has r
eleased its equipment.
1224814857 -> 110b[CMotorCall::unRegister] Remove eqt : 1036 from the map.
1224814857 -> [CMotorCallManager::onIncomingEvent] state = TERMINATED_STATE.
1224814857 -> [CMotorCallManager::onIncomingEvent] killSession.
1224814857 -> [CMotorCallManager::onNotifyCallClosed] Call : 44a44fd5b99faeb43d1
49013fe1ff30b@203.36.12.10 is closed.
1224814857 -> [CMotorCallManager::onNotifyCallClosed] Remove eqt from the map.
1224814857 -> RECEIVE MESSAGE FROM NETWORK (203.36.12.10:4135 [UDP])
----------------------utf8-----------------------
ACK sip:0437407840@203.36.12.10;user=phone SIP/2.0
Call-ID: 44a44fd5b99faeb43d149013fe1ff30b@203.36.12.10
From: "0290083901" <sip:+0290083901@203.36.12.10>;tag=759545f456b0f992e60dc0549a
a14bff
To: <sip:0437407840@203.36.12.10;user=phone>;tag=491dfd12089985ac4c9f9562fb8ac06
4
Via: SIP/2.0/UDP 203.36.12.10;branch=z9hG4bK90f251b0f06434497eb3274cfd5b613a
CSeq: 292214136 ACK
Content-Length: 0

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

1224814857 -> [CMotorCallManager::onNotifyCallClosed] Call : 44a44fd5b99faeb43d1
49013fe1ff30b@203.36.12.10 is closed.
1224814857 -> [CMotorCallManager::onNotifyCallClosed] Remove eqt from the map.
1224814859 -> 110c[CMotorCall::~CMotorCall] Call : 44a44fd5b99faeb43d149013fe1ff
30b@203.36.12.10 is being destructed.
1224814859 -> 110b[CMotorCall::~CMotorCall] Call : 44a44fd5b99faeb43d149013fe1ff
30b@203.36.12.10 is being destructed.

Thanks
patty
Member
Posts: 58
Joined: 26 Oct 2007 17:33

Re: Trouble making and recieving calls on SIP

Post by patty »

Hi,

Each time I have faced a "not acceptable here", it had nothing to do with a SIP issue, it was the call handling which refused the call (like public category issue or trunk connection category, entry not existing in real discriminator...)
alcatelguru
Member
Posts: 29
Joined: 04 Mar 2008 19:16

Re: Trouble making and recieving calls on SIP

Post by alcatelguru »

Hello,
I checked the Public and trunk COS and they all look good.Still facing the problem.

Thanks
User avatar
cavagnaro
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Re: Trouble making and recieving calls on SIP

Post by cavagnaro »

:lol: guru nothing....
I also sure that it is the public cos, why you say it is ok?
Ignorance is not the problem, the problem is the one who doesn't want to learn

OTUC/ICS ACFE/ACSE R3.0/4.0/5.0/6.0
Certified Genesys CIV 8.5
Certified Genesys Troubleshooting 8.5
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Genesys Developer
alcatelguru
Member
Posts: 29
Joined: 04 Mar 2008 19:16

Re: Trouble making and recieving calls on SIP

Post by alcatelguru »

Hello Cavagnaro,
I also checked the COS but cant locate where I am wrong.Can you help me please.

Thanks
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