Displaying 6 results from an estimated 6 matches for "q931pdu".
2005 Mar 03
0
I have met a message : "No one is available to answer at this time".
...MasterSlaveDetermination
1:21:35.134 H225 Caller:834f9d0 h323pdu.cxx(494) H245
Sending PDU:
request masterSlaveDetermination {
terminalType = 50
statusDeterminationNumber = 4068038
}
1:21:35.143 H225 Caller:834f9d0 h323pdu.cxx(494) H225
Sending PDU:
{
q931pdu = {
protocolDiscriminator = 8
callReference = 8176
from = originator
messageType = Setup
IE: Bearer-Capability = {
80 90 a5 ...
}
IE: Display = {
75 6e 6b 6e 6f 77 6e 00 un...
2005 May 23
1
OH323 CONTROL PROTOCOL ERROR
...}
}
}
}
5:59.639 ThreadID=0x4cb6a1c0 H245 Sending MasterSlaveDetermination
5:59.640 ThreadID=0x4cb6a1c0 H245 Sending PDU:
request masterSlaveDetermination {
terminalType = 60
statusDeterminationNumber = 4019430
}
5:59.641 ThreadID=0x4cb6a1c0 H225 Sending PDU:
{
q931pdu = {
protocolDiscriminator = 8
callReference = 422
from = destination
messageType = Connect
IE: Bearer-Capability = {
80 90 a5 ...
}
IE: Display = {
32 31 33 2e 32 35 35 2e 31 39 38 2e 31 31 33 00...
2005 Jan 28
3
reason 24 (Call ended with Q.931 cause)
Hi Michael and Everyone
I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting
this error
"reason 24 (Call ended with Q.931 cause)"
I've checked the Asterisk wiki and several other resources. Please can
anyone give me a hint on what the problem is I reach my wits end. Thanks
Tola
my config and debug
Configuration of OpenH323 channel driver
2005 Jan 06
0
H.323 to SIP extension
...for ip$10.0.0.5:1447/1471 set to EndedByTransportFail
0:26.546 H225 Answer:9a33350 h323.cxx(1558) H225
Sending release complete PDU: callRef=1471
-- Sending RELEASE COMPLETE
0:26.550 H225 Answer:9a33350 h323pdu.cxx(517) H225
Sending PDU:
{
q931pdu = {
protocolDiscriminator = 8
callReference = 1471
from = destination
messageType = ReleaseComplete
IE: User-User = {
25 c0 06 00 08 91 4a 00 04 58 08 11 00 ba 8b ea %.....J..X......
28 6d 5e d9 11 8d 2c 00 0b 6a 59 ee 0c 02 80 01 (m^...,..jY........
2004 Jun 29
5
SIP->Asterisk->GnuGK->Cisco 5300
Hi all,
I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300
to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can
somebody help me?
Ganbaa
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2003 Jul 21
4
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