Re: [Asterisk-Users] H323 call dropped when answered
--- "Krystian.Filiks" <[EMAIL PROTECTED]> wrote:
> did you ever get the chan_h323 working?
No
> Asterisk . wrote:
>
> >Hello,
> >
> >I posted mails regarding the same issue just 1 week back (Sub: H323 Call Dropping,
> >and then
> OH323
> >Call Dropping). I was trying to connect from my CISCO ATA to Nextone using Asterisk
> >and the
> calls
> >were dropping immediately after the calls were answered. I used chan_h323 for about
> >4 days, but
> >could not make it. Then i changed to chan_oh323 and finally got it working after
> >trying that
> for
> >another 3 days using g729 codec. I also had issues with g711, and g723. I think
> >your problem is
> >codec. Try SIP debug also and see the packets. If nothing works, try using
> >chan_oh323.
> >
> >
> >
> >>"Executing Dial("SIP/sj1-4ff7", "H323/h") in new stack"
> >>
> >>
> >
> >This one is really frustrating. I had no clue when it happened to me, and i had no
> >hangup
> command
> >in my dialplan.
> >
> >Good Luck!
> >
> >Girish
> >
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Re: [Asterisk-Users] H323 call dropped when answered
aaa
Thanks for that.
That clears things up in my head a little.
To change to oh323 do I have to recompile the * CVS sources without the
chan_h323 or can I just install oh323 abnd remove the chan_h323 from
the module directory?
What was the problem with oh323, was it just a config problem, can you
give me a pointer hov to configure it?
Did g711 and g732 finaly work with oh323?
how did you get the "Executing Dial("SIP/sj1-4ff7", "H323/h") in new
stack" to go away?
I need all the help I can get, this is my first asterisk.
Newb. warning :-)
Thanks
Krystian
Asterisk . wrote:
Hello,
I posted mails regarding the same issue just 1 week back (Sub: H323 Call Dropping, and then OH323
Call Dropping). I was trying to connect from my CISCO ATA to Nextone using Asterisk and the calls
were dropping immediately after the calls were answered. I used chan_h323 for about 4 days, but
could not make it. Then i changed to chan_oh323 and finally got it working after trying that for
another 3 days using g729 codec. I also had issues with g711, and g723. I think your problem is
codec. Try SIP debug also and see the packets. If nothing works, try using chan_oh323.
"Executing Dial("SIP/sj1-4ff7", "H323/h") in new stack"
This one is really frustrating. I had no clue when it happened to me, and i had no hangup command
in my dialplan.
Re: [Asterisk-Users] H323 call dropped when answered
did you ever get the chan_h323 working?
Asterisk . wrote:
Hello,
I posted mails regarding the same issue just 1 week back (Sub: H323 Call Dropping, and then OH323
Call Dropping). I was trying to connect from my CISCO ATA to Nextone using Asterisk and the calls
were dropping immediately after the calls were answered. I used chan_h323 for about 4 days, but
could not make it. Then i changed to chan_oh323 and finally got it working after trying that for
another 3 days using g729 codec. I also had issues with g711, and g723. I think your problem is
codec. Try SIP debug also and see the packets. If nothing works, try using chan_oh323.
"Executing Dial("SIP/sj1-4ff7", "H323/h") in new stack"
This one is really frustrating. I had no clue when it happened to me, and i had no hangup command
in my dialplan.
Good Luck!
Girish
--- "Krystian.Filiks" <[EMAIL PROTECTED]> wrote:
Hi,
This is the scenario
I have the SJlabs phone with g711ulaw active and the rest disabled.
I have * with chan_h323
I have a Quintum DX that supports, g723.1 , g729AB, ulaw and alaw.
The problem is that, it does not mather what I put in the
extensions.conf I have tried all possible ways that I so far could find
using the net.
I tried all possible codecs ulaw, alaw, g723 and g729 always the same
result.
The phone rings but as soon as answered it dissconnects.
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Re: [Asterisk-Users] H323 call dropped when answered
Hello,
I posted mails regarding the same issue just 1 week back (Sub: H323 Call Dropping, and
then OH323
Call Dropping). I was trying to connect from my CISCO ATA to Nextone using Asterisk
and the calls
were dropping immediately after the calls were answered. I used chan_h323 for about 4
days, but
could not make it. Then i changed to chan_oh323 and finally got it working after
trying that for
another 3 days using g729 codec. I also had issues with g711, and g723. I think your
problem is
codec. Try SIP debug also and see the packets. If nothing works, try using chan_oh323.
> "Executing Dial("SIP/sj1-4ff7", "H323/h") in new stack"
This one is really frustrating. I had no clue when it happened to me, and i had no
hangup command
in my dialplan.
Good Luck!
Girish
--- "Krystian.Filiks" <[EMAIL PROTECTED]> wrote:
> Hi,
> This is the scenario
> I have the SJlabs phone with g711ulaw active and the rest disabled.
> I have * with chan_h323
> I have a Quintum DX that supports, g723.1 , g729AB, ulaw and alaw.
>
> The problem is that, it does not mather what I put in the
> extensions.conf I have tried all possible ways that I so far could find
> using the net.
> I tried all possible codecs ulaw, alaw, g723 and g729 always the same
> result.
> The phone rings but as soon as answered it dissconnects.
>
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Re: [Asterisk-Users] H323 call dropped when answered
Hi,
This is the scenario
I have the SJlabs phone with g711ulaw active and the rest disabled.
I have * with chan_h323
I have a Quintum DX that supports, g723.1 , g729AB, ulaw and alaw.
The problem is that, it does not mather what I put in the
extensions.conf I have tried all possible ways that I so far could
find using the net.
I tried all possible codecs ulaw, alaw, g723 and g729 always the same
result.
The phone rings but as soon as answered it dissconnects.
The debug shows
--
Executing Dial("SIP/sj1-4ff7",
"H323/0797617729") in new stack
--
Called 0797617729
--
H323/0797617729 is ringing
--
H323/0797617729 answered SIP/sj1-4ff7
==
Spawn extension (default, 0797617729, 1) exited non-zero on
'SIP/sj1-4ff7'
-- Executing Dial("SIP/sj1-4ff7",
"H323/h") in new stack
--
Called h
==
Spawn extension (default, h, 1) exited non-zero on 'SIP/sj1-4ff7'
The first
Dial is normal but the 2nd Dial “Executing Dial("SIP/sj1-4ff7", "H323/h") in
new stack”
Where do
that come from?
PLEASE someone HELP!
The * have the config below
In extensions.conf I use
[globals]
[default]
exten => _.,1,Dial(H323/${EXTEN})
in H323.conf I use
[general]
port = 1720
bindaddr = 195.216.65.212
tos=lowdelay
allow=all
gatekeeper = 195.216.65.215
AllowGKRouted = yes
context=default
[AST37]
type=h323
In SIP.conf I have
[general]
port=5060
bindaddr=xxx.xxx.xxx.xxx
[sj1]
type=friend
context=default
host=dynamic
disallow=all
allow=all
username=sj1
secret=sj1
[sj2]
type=friend
context=default
host=xxx.xxx.xxx.xxx
allow=ulaw
username=sj1
secret=sj1
;;
administrator tootai wrote:
Krystian
Filiks a écrit :
Like you suggested I tried the g.711 now and
got the same, The called number rings but when answered it dropped.
I connect to a Quintum Tenor DX.
The part I'm curious about is 6:53.985
Transactor:8140ee8 h323trans.cxx(678) Trans admissionRequest
rejected: requestDenied
6:53.988 H225 Caller:8159198 h323.cxx(2660) H225
Gatekeeper refused admission: requestDenied
6:53.959 H225 Caller:813c890 h323pdu.cxx(1159) H225
Read error (0):
Does anyone have a clue where to look for the problem?
here is a trace,
-- Executing Dial("SIP/sj1-a7e9", "H323/[EMAIL PROTECTED]") in new stack
Allowed Codecs:
Table:
G.711-uLaw-64k{sw} <1>
Set:
0:
0:
G.711-uLaw-64k{sw} <1>
-- Making call to [EMAIL PROTECTED] using gatekeeper.
channelsOpen = 1
channelsOpen = 0
6:53.959 H225 Caller:813c890 h323pdu.cxx(1159) H225
Read error (0):
== New H.323 Connection created.
-- sj1 is calling host [EMAIL PROTECTED]
-- Call token is ip$localhost/31767
-- Call reference is 31767
-- Called [EMAIL PROTECTED]
-- ClearCall: Request to clear call with token
ip$localhost/31767
-- Sending RELEASE COMPLETE
== Spawn extension (default, h, 1) exited non-zero on 'SIP/sj1-a7e9'
6:53.985 Transactor:8140ee8 h323trans.cxx(678) Trans
admissionRequest rejected: requestDenied
6:53.988 H225 Caller:8159198 h323.cxx(2660) H225
Gatekeeper refused admission: requestDenied
6:54.004 H323 Cleaner h323.cxx(1542) H323
Connection ip$localhost/31766 terminated.
-- Call with Tenor Gateway [195.216.65.215] completed
(EndedByLocalUser)
== H.323 Connection deleted.
What's [EMAIL PROTECTED]? If you are register to GK
H323/ is enough. I don't understand your h EP. And
also request denied seems that you need to register. But I don't know
how work Quintum, maybe I'm wrong.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
administrator tootai
Sent: Thursday, August 12, 2004 6:02 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H323 call dropped when answered
Krystian.Filiks a écrit :
Hello anyone that can help me here?? please
read below.
[...]
Allowed Codecs:
Table:
G.723.1{sw} <1>
Set:
0:
0:
G.723.1{sw} <1>
G.723.1 is not a codec in * Use g711 instead. If your GK is GnuGK, see
his debug logs. Also, run aste
Re: [Asterisk-Users] H323 call dropped when answered
Krystian Filiks a écrit :
Like you suggested I tried the g.711 now and got the same, The called number rings but
when answered it dropped.
I connect to a Quintum Tenor DX.
The part I'm curious about is
6:53.985 Transactor:8140ee8h323trans.cxx(678) Trans admissionRequest rejected: requestDenied
6:53.988 H225 Caller:8159198 h323.cxx(2660) H225Gatekeeper refused admission: requestDenied
6:53.959 H225 Caller:813c890 h323pdu.cxx(1159) H225Read error (0):
Does anyone have a clue where to look for the problem?
here is a trace,
-- Executing Dial("SIP/sj1-a7e9", "H323/[EMAIL PROTECTED]") in new stack
Allowed Codecs:
Table:
G.711-uLaw-64k{sw} <1>
Set:
0:
0:
G.711-uLaw-64k{sw} <1>
-- Making call to [EMAIL PROTECTED] using gatekeeper.
channelsOpen = 1
channelsOpen = 0
6:53.959 H225 Caller:813c890 h323pdu.cxx(1159) H225Read error (0):
== New H.323 Connection created.
-- sj1 is calling host [EMAIL PROTECTED]
-- Call token is ip$localhost/31767
-- Call reference is 31767
-- Called [EMAIL PROTECTED]
-- ClearCall: Request to clear call with token ip$localhost/31767
-- Sending RELEASE COMPLETE
== Spawn extension (default, h, 1) exited non-zero on 'SIP/sj1-a7e9'
6:53.985 Transactor:8140ee8h323trans.cxx(678) Trans admissionRequest rejected: requestDenied
6:53.988 H225 Caller:8159198 h323.cxx(2660) H225Gatekeeper refused admission: requestDenied
6:54.004 H323 Cleaner h323.cxx(1542) H323Connection ip$localhost/31766 terminated.
-- Call with Tenor Gateway [195.216.65.215] completed (EndedByLocalUser)
== H.323 Connection deleted.
What's [EMAIL PROTECTED] If you are register to GK H323/ is
enough. I don't understand your h EP. And also request denied seems that
you need to register. But I don't know how work Quintum, maybe I'm wrong.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of administrator tootai
Sent: Thursday, August 12, 2004 6:02 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H323 call dropped when answered
Krystian.Filiks a écrit :
Hello anyone that can help me here?? please read below.
[...]
Allowed Codecs:
Table:
G.723.1{sw} <1>
Set:
0:
0:
G.723.1{sw} <1>
G.723.1 is not a codec in * Use g711 instead. If your GK is GnuGK, see
his debug logs. Also, run asteriks in debug mode and check logs in full
file.
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RE: [Asterisk-Users] H323 call dropped when answered
Can anyone tell me what this means? 0:54.517 H225 Caller:8141ae8 h323pdu.cxx(1213) H225 Write PDU failed (32): Broken pipe and why this might happen, my call is dropped just after receiving this Thanks /Krystian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 call dropped when answered
Like you suggested I tried the g.711 now and got the same, The called number rings but
when answered it dropped.
I connect to a Quintum Tenor DX.
The part I'm curious about is
6:53.985 Transactor:8140ee8h323trans.cxx(678) Trans
admissionRequest rejected: requestDenied
6:53.988 H225 Caller:8159198 h323.cxx(2660) H225Gatekeeper
refused admission: requestDenied
6:53.959 H225 Caller:813c890 h323pdu.cxx(1159) H225Read error (0):
Does anyone have a clue where to look for the problem?
here is a trace,
-- Executing Dial("SIP/sj1-a7e9", "H323/[EMAIL PROTECTED]") in new stack
Allowed Codecs:
Table:
G.711-uLaw-64k{sw} <1>
Set:
0:
0:
G.711-uLaw-64k{sw} <1>
-- Making call to [EMAIL PROTECTED] using gatekeeper.
channelsOpen = 1
channelsOpen = 0
6:53.959 H225 Caller:813c890 h323pdu.cxx(1159) H225Read error (0):
== New H.323 Connection created.
-- sj1 is calling host [EMAIL PROTECTED]
-- Call token is ip$localhost/31767
-- Call reference is 31767
-- Called [EMAIL PROTECTED]
-- ClearCall: Request to clear call with token ip$localhost/31767
-- Sending RELEASE COMPLETE
== Spawn extension (default, h, 1) exited non-zero on 'SIP/sj1-a7e9'
6:53.985 Transactor:8140ee8h323trans.cxx(678) Trans
admissionRequest rejected: requestDenied
6:53.988 H225 Caller:8159198 h323.cxx(2660) H225Gatekeeper
refused admission: requestDenied
6:54.004 H323 Cleaner h323.cxx(1542) H323Connection
ip$localhost/31766 terminated.
-- Call with Tenor Gateway [195.216.65.215] completed (EndedByLocalUser)
== H.323 Connection deleted.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of administrator tootai
Sent: Thursday, August 12, 2004 6:02 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H323 call dropped when answered
Krystian.Filiks a écrit :
> Hello anyone that can help me here?? please read below.
> [...]
>
>> Allowed Codecs:
>>
>> Table:
>>
>>G.723.1{sw} <1>
>>
>> Set:
>>
>>0:
>>
>> 0:
>>
>>G.723.1{sw} <1>
>>
G.723.1 is not a codec in * Use g711 instead. If your GK is GnuGK, see
his debug logs. Also, run asteriks in debug mode and check logs in full
file.
--
Daniel
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Re: [Asterisk-Users] H323 call dropped when answered
Krystian.Filiks a écrit :
Hello anyone that can help me here?? please read below.
[...]
Allowed Codecs:
Table:
G.723.1{sw} <1>
Set:
0:
0:
G.723.1{sw} <1>
G.723.1 is not a codec in * Use g711 instead. If your GK is GnuGK, see
his debug logs. Also, run asteriks in debug mode and check logs in full
file.
--
Daniel
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Re: [Asterisk-Users] H323 call dropped when answered
Hello anyone that can help me here?? please read below.
Regards
Krystian
Krystian Filiks wrote:
Hi All.
I’m using RedHat 9
I configured the
chan_h323 and asterisk from CVS.
This is the scenario SJ_lab_phone(sip)
-à Asterisk ---à H323 GK à PSTN
I have tried all codec’s
and always the same result, the called phone will ring without dropping
for how ever I allow it to but as soon as it is answered it immediately
gets disconnected.
Have anyone got a clue
where to look for the problem?
Here is a Debug trace:
-- Executing Dial("SIP/sj1-6a47",
"H323/[EMAIL PROTECTED]") in
new stack
Allowed Codecs:
Table:
G.723.1{sw} <1>
Set:
0:
0:
G.723.1{sw} <1>
--
Making call to [EMAIL PROTECTED] using gatekeeper.
==
New H.323 Connection created.
--sj1 is calling host [EMAIL PROTECTED]
--
Call token is ip$localhost/28087
--
Call reference is 28087
--
Called [EMAIL PROTECTED]
1:56.153 H225 Caller:813bbe0 h323trans.cxx(656) Trans Timeout
on request seqnum=14213, try
#1 of 2
1:59.163 H225 Caller:813bbe0 h323trans.cxx(656) Trans Timeout
on request seqnum=14213, try
#2 of 2
--
Sending SETUP message
--
Received Facility message...
--
Received Facility message...
--
Received Facility message...
--
Received Facility message...
--
Received Facility message...
=*=
In CreateRealTimeLogicalChannel for call 28087
--externalIpAddress: xxx.xxx.xxx.xxx
--externalPort: 15702
--SessionID: 1
--
Direction: IsReceiver
--
Started logical channel: receiving G.723.1{sw}
--channelsOpen = 1
--
Ringing phone for "xxx.xxx.xxx.xxx"
--
H323/xxx.xxx.xxx.xxx is ringing
2:10.228 H225 Caller:813bbe0 h323.cxx(2898) H225 Received
connect PDU.
=*=
In CreateRealTimeLogicalChannel for call
28087
--externalIpAddress: xxx.xxx.xxx.xxx
--externalPort: 15702
--SessionID: 1
--
Direction: IsTransmitter
--
Started logical channel: sending G.723.1{sw}
--channelsOpen = 2
--
Connection Established with "Tenor Gateway [xxx.xxx.xxx.xxx]"
--
H323/xxx.xxx.xxx.xxx answered SIP/sj1-6a47
--
Received Facility message...
--ClearCall: Request to clear call with token
ip$localhost/28087
--
Sending RELEASE COMPLETE
==
Spawn extension (default, 0797617729, 1) exited non-zero on
'SIP/sj1-6a47'
--
Executing Dial("SIP/sj1-6a47",
"H323/[EMAIL PROTECTED]") in new stack
Allowed Codecs:
Table:
G.723.1{sw} <1>
Set:
0:
0:
G.723.1{sw} <1>
channelsOpen = 1
--
Making call to [EMAIL PROTECTED] using
gatekeeper.
channelsOpen = 0
2:10.385 H225 Caller:813bbe0 h323pdu.cxx(1159) H225 Read err
or (0):
==
New H.323 Connection created.
-- sj1 is calling host [EMAIL PROTECTED]
--
Call token is ip$localhost/28088
--
Call reference is 28088
--
Called [EMAIL PROTECTED]
--ClearCall: Request to clear call with token
ip$localhost/28088
--
Sending RELEASE COMPLETE
==
Spawn extension (default, h, 1) exited non-zero on 'SIP/sj1-6a47'
2:10.404 Transactor:8140c30 h323trans.cxx(678) Trans admissio
nRequest rejected: requestDenied
2:10.406 H225 Caller:8152bb8 h323.cxx(2660) H225 Gatekeep
er refused admission: requestDenied
2:10.423 H323 Cleaner h323.cxx(1542) H323 Connecti
on ip$localhost/28087
terminated.
--
Call with Tenor Gateway [xxx.xxx.xxx.xxx]
completed (EndedByLocalUser)
==
H.323 Connection deleted.
2:10.431 H323 Cleaner h323.cxx(1542) H323 Connecti
on ip$localhost/28088
terminated.
--
Call with h completed (EndedByLocalUser)
==
H.323 Connection deleted.
[Asterisk-Users] H323 call dropped when answered
Hi All.
I’m using RedHat 9
I configured the chan_h323 and asterisk from CVS.
This is the scenario SJ_lab_phone(sip) -à Asterisk
---à H323 GK
à PSTN
I have tried all codec’s and always the same result,
the called phone will ring without dropping for how ever I allow it to but as
soon as it is answered it immediately gets disconnected.
Have anyone got a clue where to look for the problem?
Here is a Debug trace:
-- Executing Dial("SIP/sj1-6a47",
"H323/[EMAIL PROTECTED]") in
new stack
Allowed Codecs:
Table:
G.723.1{sw} <1>
Set:
0:
0:
G.723.1{sw} <1>
-- Making call
to [EMAIL PROTECTED] using gatekeeper.
==
New H.323 Connection created.
-- sj1 is calling host [EMAIL PROTECTED]
--
Call token is ip$localhost/28087
--
Call reference is 28087
--
Called [EMAIL PROTECTED]
1:56.153
H225 Caller:813bbe0 h323trans.cxx(656) Trans Timeout
on request
seqnum=14213, try #1 of 2
1:59.163
H225 Caller:813bbe0 h323trans.cxx(656) Trans Timeout
on request
seqnum=14213, try #2 of 2
--
Sending SETUP message
--
Received Facility message...
--
Received Facility message...
--
Received Facility message...
--
Received Facility message...
--
Received Facility message...
=*=
In CreateRealTimeLogicalChannel for call 28087
-- externalIpAddress:
xxx.xxx.xxx.xxx
-- externalPort:
15702
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.723.1{sw}
-- channelsOpen = 1
--
Ringing phone for "xxx.xxx.xxx.xxx"
--
H323/xxx.xxx.xxx.xxx is ringing
2:10.228
H225 Caller:813bbe0
h323.cxx(2898) H225 Received
connect PDU.
=*=
In CreateRealTimeLogicalChannel for call 28087
-- externalIpAddress:
xxx.xxx.xxx.xxx
-- externalPort:
15702
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.723.1{sw}
-- channelsOpen = 2
--
Connection Established with "Tenor Gateway [xxx.xxx.xxx.xxx]"
--
H323/xxx.xxx.xxx.xxx answered SIP/sj1-6a47
--
Received Facility message...
-- ClearCall: Request to clear call with token
ip$localhost/28087
--
Sending RELEASE COMPLETE
== Spawn
extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-6a47'
--
Executing Dial("SIP/sj1-6a47", "H323/[EMAIL PROTECTED]")
in new stack
Allowed Codecs:
Table:
G.723.1{sw} <1>
Set:
0:
0:
G.723.1{sw} <1>
channelsOpen = 1
-- Making call
to [EMAIL PROTECTED] using gatekeeper.
channelsOpen = 0
2:10.385
H225 Caller:813bbe0
h323pdu.cxx(1159) H225 Read err
or (0):
==
New H.323 Connection created.
-- sj1 is
calling host [EMAIL PROTECTED]
--
Call token is ip$localhost/28088
--
Call reference is 28088
--
Called [EMAIL PROTECTED]
-- ClearCall: Request to clear call with token
ip$localhost/28088
--
Sending RELEASE COMPLETE
== Spawn
extension (default, h, 1) exited non-zero on 'SIP/sj1-6a47'
2:10.404
Transactor:8140c30 h323trans.cxx(678) Trans admissio
nRequest rejected: requestDenied
2:10.406
H225 Caller:8152bb8
h323.cxx(2660) H225 Gatekeep
er refused
admission: requestDenied
2:10.423
H323 Cleaner
h323.cxx(1542) H323 Connecti
on ip$localhost/28087
terminated.
-- Call with
Tenor Gateway [xxx.xxx.xxx.xxx] completed (EndedByLocalUser)
==
H.323 Connection deleted.
2:10.431
H323 Cleaner
h323.cxx(1542) H323 Connecti
on ip$localhost/28088
terminated.
-- Call with h
completed (EndedByLocalUser)
==
H.323 Connection deleted.
