Martin Pycko
2003-Jul-21 10:25 UTC
[Asterisk-Users] anyone with X100P & Callerid working outside US ?
I'm just curious if anyone has the X100P & Callerid receiving working outside US. Replies are appreciated. Also if it's not working for you in a certain coutry you can respond too. regards Martin
Dan
2003-Jul-21 10:48 UTC
[Asterisk-Users] anyone with X100P & Callerid working outside US ?
Hi Martin, For me it just display my own PSTN number, extracted as caller id from the PSTN line. I am located in Romania. Best regards, Dan P.S. Some analog phones with internal caller id displays the same number, but others (especially some Siemens ones) display the correct caller id. I think that the X100P card does not extract the correct part of the callerid information. ----- Original Message ----- From: "Martin Pycko" <martinp@digium.com> To: <asterisk-users@lists.digium.com> Sent: Monday, July 21, 2003 8:25 PM Subject: [Asterisk-Users] anyone with X100P & Callerid working outside US ?> I'm just curious if anyone has the X100P & Callerid receiving working > outside US. > > Replies are appreciated. Also if it's not working for you in a certain > coutry you can respond too. > > regards > Martin > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Tamas Levente
2003-Jul-21 13:21 UTC
[Asterisk-Users] anyone with X100P & Callerid working outside US ?
How can I use ztmonitor to figure out the caller id sent by the telco? Because it is not working for me in Chicago. ----- Original Message ----- From: "Martin Pycko" <martinp@digium.com> To: <levi@televersions.com> Sent: Monday, July 21, 2003 9:03 PM Subject: Re: [Asterisk-Users] anyone with X100P & Callerid working outside US ?> It's possible that your telco first transmits the DID (your number) and > then later on the callerid ... > > Did you "listen" for it with ztmonitor ? .... If my suspicion is right ? > > regards > Martin > > On Mon, 21 Jul 2003, Dan wrote: > > > Hi Martin, > > > > For me it just display my own PSTN number, extracted as caller id fromthe> > PSTN line. > > I am located in Romania. > > > > Best regards, > > Dan > > P.S. Some analog phones with internal caller id displays the samenumber,> > but others (especially some Siemens ones) display the correct caller id. > > I think that the X100P card does not extract the correct part of the > > callerid information. > > > > > > ----- Original Message ----- > > From: "Martin Pycko" <martinp@digium.com> > > To: <asterisk-users@lists.digium.com> > > Sent: Monday, July 21, 2003 8:25 PM > > Subject: [Asterisk-Users] anyone with X100P & Callerid working outsideUS ?> > > > > > > I'm just curious if anyone has the X100P & Callerid receiving working > > > outside US. > > > > > > Replies are appreciated. Also if it's not working for you in a certain > > > coutry you can respond too. > > > > > > regards > > > Martin > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Armand A. Verstappen
2003-Jul-21 17:54 UTC
[Asterisk-Users] anyone with X100P & Callerid working outside US ?
On Mon, 2003-07-21 at 19:25, Martin Pycko wrote:> I'm just curious if anyone has the X100P & Callerid receiving working > outside US.It does not work in the Netherlands. The Netherlands does not use FSK signalling, but DTMF signalling: 1) polarity reversal 2) DTMF: <D><Number><C> 3) ring signal where <D> and <C> are the DTMF tones 'D' and 'C' respectively, signalling start and end of DTMF Caller-ID transfer. Exact specification (including length of tones and pauses) is in this document: http://www.kpn.com/common/downloads/01_Part2-PSTN_V32.pdf paragraph 6.2.3. At least Sweden and Denmark use very similar CLIP protocols, the difference being mainly in the start and end tones used. The different protocol also bites on the other end, as asterisk will send callerid information to a phone connected to a TDM40B for example using the FSK protocol. Dutch phones don't understand FSK, and hence don't pick up on the caller id. I'm very interested in solving this problem, as it makes asterisk only usable in the Netherlands using ISDN BRI or PRI on the PSTN side, and imported phones on the (analog) internal side. I just don't have any idea where in the source, and how... One possible solution for the 'inside' problem: There's one company in the Netherlands offering telefony over cable infrastructure, they use FSK signalling for Caller-ID presentation. I'll get my hands on a phone suited for their network soon, wich will allow me to verify if they work with asterisk. -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 armand@nl.envida.net 3531 AH Utrecht tel: +31 (0)30 298 2255 Postbus 19127 fax: +31 (0)30 298 2111 3501 DC Utrecht -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030721/b8d17a59/attachment.pgp
I got my license yesterday and I'm trying to make it work
with my Quicknet Linejack.
The last lines of asterisk's "h.323 trace 5" are:
0:56.570 H245:8113c78 h323neg.cxx(620) H245
Received TerminalCapabilitySetAck: state=InProgress pduSeq=2 outSeq=2
0:56.571 H245:8113c78 h323neg.cxx(630) H245
TerminalCapabilitySet Sent.
0:56.571 H245:8113c78 h323.cxx(3876) H323
InternalEstablishedConnectionCheck:
connectionState=HasExecutedSignalConnect fastStartState=FastStartDisabled
0:56.571 H245:8113c78 h323.cxx(3163) H245
Received TPKT: size=2 pos=0.0 {
20 a0
}
0:56.572 H245:8113c78 h323pdu.cxx(474) H245
Receiving PDU:
response masterSlaveDeterminationAck {
decision = slave <<null>>
}
0:56.572 H245:8113c78 h323neg.cxx(406) H245
Received MasterSlaveDeterminationAck: state=Incoming
0:56.572 H245:8113c78 h323.cxx(3876) H323
InternalEstablishedConnectionCheck:
connectionState=HasExecutedSignalConnect fastStartState=FastStartDisabled
0:56.572 H245:8113c78 h323caps.cxx(1699) H323
FindCapability: "T.120"
0:56.573 H245:8113c78 h323.cxx(3933) H245
Default OnSelectLogicalChannels, FastStartDisabled
0:56.573 H245:8113c78 h323caps.cxx(1735) H323
FindCapability: G.729A{n/a} <1>
0:56.573 H245:8113c78 h323.cxx(3163) H245
Received TPKT: size=2 pos=0.0 {
4a 40 J@
}
0:56.574 H245:8113c78 h323pdu.cxx(474) H245
Receiving PDU:
command endSessionCommand disconnect <<null>>
0:56.574 H245:8113c78 h323ep.cxx(1537) H323
Clearing connection ip$localhost/24778 reason=EndedByRemoteUser
0:56.574 H245:8113c78 h323.cxx(1403) H323 Call
end reason for ip$localhost/24778 set to EndedByRemoteUser
0:56.574 H245:8113c78 h323.cxx(1421) H225
Sending release complete PDU: callRef=24778
0:56.576 H245:8113c78 h323pdu.cxx(474) H245
Sending PDU:
command endSessionCommand disconnect <<null>>
0:56.577 H245:8113c78 h323pdu.cxx(474) H225
Sending PDU:
{
q931pdu = {
protocolDiscriminator = 8
callReference = 24778
from = originator
messageType = ReleaseComplete
IE: Cause - Normal call clearing = {
80 90 ..
}
IE: User-User = {
25 c0 06 00 08 91 4a 00 03 58 58 00 11 00 34 d6 %.....J..XX...4.
f0 ea fd ba d7 11 9c 2c ca 60 f5 15 2c 7c 08 80 .......,.`..,|..
01 00 ..
}
}
h225pdu = {
h323_uu_pdu = {
h323_message_body = releaseComplete {
protocolIdentifier = 0.0.8.2250.0.3
reason = undefinedReason <<null>>
callIdentifier = {
guid = 16 octets {
34 d6 f0 ea fd ba d7 11 9c 2c ca 60 f5 15 2c 7c 4........,.`..,|
}
}
}
h245Tunneling = FALSE
}
}
}
0:56.592 H225 Caller:8101830 h323.cxx(1620) H225
Handling PDU: ReleaseComplete callRef=24778
0:56.592 H225 Caller:8101830 h323ep.cxx(1537) H323
Clearing connection ip$localhost/24778 reason=EndedByTransportFail
0:56.592 H225 Caller:8101830 h323.cxx(1610) H225
Signal channel closed.
0:56.593 H225 Caller:8101830 tlibthrd.cxx(1072) PWLib
Ended thread 0x8101830 H225 Caller:8101830
0:57.038 H323 Cleaner transports.cxx(1048) H323
H323Transport::CleanUpOnTermination for H245:8113c78
0:57.038 H323 Cleaner tlibthrd.cxx(672) PWLib
Destroyed thread 0x8113c78 H245:8113c78
0:57.038 H323 Cleaner transports.cxx(966) H323
H323Transport::Close
0:57.473 H323 Cleaner transports.cxx(1048) H323
H323Transport::CleanUpOnTermination for H225 Caller:8101830
0:57.474 H323 Cleaner tlibthrd.cxx(672) PWLib
Destroyed thread 0x8101830 H225 Caller:8101830
0:57.474 H323 Cleaner h323.cxx(1518) H323
Connection ip$localhost/24778 terminated.
0:57.474 H323 Cleaner h323.cxx(1353) H323
Connection ip$localhost/24778 deleted.
0:57.474 H323 Cleaner h323ep.cxx(1594) H323
Cleaning up connections
-- H323/Hermann answered Phone/phone0
WARNING[15376]: File res_parking.c, Line 209 (ast_bridge_call): Bridge
failed on channels Phone/phone0 and H323/Hermann
== Spawn extension (demo, s, 4) exited non-zero on 'Phone/phone0'
NOTICE[15376]: File channel.c, Line 1296 (ast_set_write_format): Unable to
find
a path from 0 to 69
-- Hungup 'Phone/phone0'
The other side is a Planet VIP-400 VOIP box(http://www.planet.com.tw)
Its logs are:
2- HSSM 2 HSMU 0: SM "WAIT TELE CONNECT" ==> "H225 WAIT
CONNECT"
0- HSSM 2 HSMU 0: SM "H225 WAIT CONNECT" <-
"H225_connect"
1- HSSM 2 HSMU 0: SM "H225 WAIT CONNECT" ==> "H245 WAIT
COMPLETE"
0- HSMU 2 HSMU 0: Product ID = "The NuFone Network's H.323 Channel
Driver for Asterisk", netmeeting 0
1- HSMU 2 HSMU 0: Version ID = "0.1.0 (OpenH323 v1.11.7)"
0- HSMU 2 HSMU 0: Starting Capabilities Exchange
21- HSMU 2 HSMU 0: set timer for 90000 in state H245 WAIT COMPLETE
430- RADH 2 HSMU RAD: cmHookInConnect
0- RADH 2 HSMU RAD: cmHookClose
2- RAD 2 HSMU 0:
cmEvCallControlStateChanged(cmControlStateTransportConnected, - )
10- RADH 2 HSMU RAD: cmHookSend(terminalCapabilitySet)
10- RADH 2 HSMU RAD: cmHookSend(masterSlaveDetermination)
17- RADH 2 HSMU RAD: cmHookRecv(terminalCapabilitySet)
8- RAD 2 HSMU 0: cmEvCallCapabilities
0- HSMU 0
0- HSMU 0 Capability Set
0- HSMU 0 [1] g729AnnexA: Audio Receive
1- RAD 2 HSMU 0: cmEvCallCapabilitiesExt
4- RADH 2 HSMU RAD: cmHookSend(terminalCapabilitySetAck)
6- RADH 2 HSMU RAD: cmHookRecv(masterSlaveDetermination)
15- RADH 2 HSMU RAD: cmHookSend(masterSlaveDeterminationAck)
11- HSMU 0 Remote capabilities list:
0- HSMU 0 [1] g729AnnexA: Audio Receive
0- HSMU 0 Try matching local element:
0- HSMU 0 [1] g7231: Audio Receive and Transmit
0- HSMU 0 Try matching local element:
0- HSMU 0 [2] g729: Audio Receive and Transmit
1- HSMU 0 Try matching local element:
0- HSMU 0 [3] g711Ulaw64k: Audio Receive and Transmit
0- HSMU 0 Try matching local element:
0- HSMU 0 [4] t38fax: Data Receive and Transmit
0- HSMU 0 Try matching local element:
0- HSMU 0 [5] g729: Audio Receive and Transmit
0- HSMU 3 HSMU 0: Capabilities: NO MATCH FOUND!
0- HSSM 2 HSMU 0: SM "H245 WAIT COMPLETE" <- "error
release"
0- HSMU 2 HSMU 0: abort timer in state H245 WAIT COMPLETE
0- HSSM 2 HSMU 0: SM "H245 WAIT COMPLETE" ==>
"RELEASING"
10- RADH 2 HSMU RAD: cmHookSend(endSessionCommand)
3- RAD 2 HSMU 0:
cmEvCallControlStateChanged(cmControlStateTransportDisconnected, - )
2- RADH 2 HSMU RAD: cmHookClose
7- RADH 2 HSMU RAD: cmHookSend(releaseComplete)
7- RAD 2 HSMU 0: cmEvCallStateChanged(State = cmCallStateDisconnected,
cmCallStateModeDisconnectedLocal) - remote
5- RAD 2 HSMU 0: cmEvCallStateChanged(State = cmCallStateIdle,
cmCallStateTransfering) - remote
2- HSSM 2 HSMU 0: release >> PSU
0- HSSM 2 HSMU 0: RELEASE: release reason 11
2- NMM 2 NMM: 0, get_tone_table_entry(Tone_id=20) Using Default tone
table
0- NMM 4 NMM: 0, states: oper=NORMAL, admin=NORMAL, call=TEAR_DOWN
0- NMM 2 NMM: 0, Call Record received '', DSP 0:0
0- NMM 2 Originated: remote
0- NMM 2 Terminated: local
1- NMM 2 Call state: TEAR_DOWN
0- NMM 2 Release reason: GG_REL_BUSY
0- NMM 2 Seized ts = 4448146 msec (since system startup)
0- NMM 2 Connected ts = 4826 msec (since line seizure)
0- NMM 2 Call duration = 6026 msec (since line seizure)
0- NMM 2 = 1200 msec (since connected ts)
0- NMM 2 Neg. coding = 0, ts = 20 msec (since line seizure)
1- HSSM 1 HSMU 0: Double release (IF)
0- HSSM 2 HSMU 0: SM "RELEASING" ==> "WAIT RELEASE
RESPONSE"
0- HSMU 2 HSMU 0: CapabilitiesExt_msg
0- HSSM 2 HSMU 0: SM "WAIT RELEASE RESPONSE" <-
"H323_disconnect"
4- RADH 2 HSMU RAD: cmHookClose
5- HSSM 2 HSMU 0: SM "WAIT RELEASE RESPONSE" <-
"H323_idle"
0- HSSM 2 HSMU 0: << release response
0- HSSM 2 HSMU 0: SM "WAIT RELEASE RESPONSE" <-
"psu_release_response"
0- HSSM 2 HSMU 0: SM "WAIT RELEASE RESPONSE" ==>
"IDLE"
2488- NMM 2 NMM: 0, get_tone_table_entry(Tone_id=20) Using Default tone
table
1- NMM 4 NMM: 0, states: oper=NORMAL, admin=NORMAL, call=TEAR_DOWN
1000- NMM 4 NMM: 0, states: oper=NORMAL, admin=NORMAL, call=IDLE
Looks like a problem in the protocol in Asterisk's end because it
receives a "reason=EndedByTransportFail".
The Planet's box is 200.221.36.67. The extension 0800781800 can be
called by anyone who wants to make a test....
or I need necessarily a Digium's Board to make it working ;-)
Isamar