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Krystian Filiks

2004-Aug-11 16:56 UTC

[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) ---------------> Asterisk
-------------> H323 GK --------------> 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",
"H323/0797617729@xxx.xxx.xxx.xxx") in
 new stack
Allowed Codecs:
         Table:
   G.723.1{sw} <1>
 Set:
   0:
     0:
       G.723.1{sw} <1>
 
 -- Making call to 0797617729@xxx.xxx.xxx.xxx using gatekeeper.
        == New H.323 Connection created.
        -- sj1 is calling host 0797617729@xxx.xxx.xxx.xxx
        -- Call token is ip$localhost/28087
        -- Call reference is 28087
    -- Called 0797617729@xxx.xxx.xxx.xxx
  1:56.153          H225 Caller:813bbe0    h323trans.cxx(656)   Trans
Timeout
on request seqnum=14213, try #1 of 2
  1:59.163          H225 Caller:813bbe0    h323trans.cxx(656)   Trans
Timeout
on request seqnum=14213, try #2 of 2
        -- Sending SETUP message
        -- Received Facility message...
        -- Received Facility message...
        -- Received Facility message...
        -- Received Facility message...
        -- Received Facility message...
        =*= In CreateRealTimeLogicalChannel for call 28087
                -- externalIpAddress: xxx.xxx.xxx.xxx
                -- externalPort: 15702
                -- SessionID: 1
                -- Direction: IsReceiver
         -- Started logical channel: receiving G.723.1{sw}
                -- channelsOpen = 1
        -- Ringing phone for "xxx.xxx.xxx.xxx"
    -- H323/xxx.xxx.xxx.xxx is ringing
  2:10.228          H225 Caller:813bbe0         h323.cxx(2898)  H225
Received
 connect PDU.
        =*= In CreateRealTimeLogicalChannel for call 28087
                -- externalIpAddress: xxx.xxx.xxx.xxx
                -- externalPort: 15702
                -- SessionID: 1
                -- Direction: IsTransmitter
         -- Started logical channel: sending G.723.1{sw}
                -- channelsOpen = 2
        -- Connection Established with "Tenor Gateway
[xxx.xxx.xxx.xxx]"
    -- H323/xxx.xxx.xxx.xxx answered SIP/sj1-6a47
        -- Received Facility message...
        -- ClearCall: Request to clear call with token
ip$localhost/28087
        -- Sending RELEASE COMPLETE
  == Spawn extension (default, 0797617729, 1) exited non-zero on
'SIP/sj1-6a47'
    -- Executing Dial("SIP/sj1-6a47",
"H323/h@xxx.xxx.xxx.xxx") in new
stack
Allowed Codecs:
         Table:
   G.723.1{sw} <1>
 Set:
   0:
     0:
       G.723.1{sw} <1>
 
                channelsOpen = 1
 -- Making call to h@xxx.xxx.xxx.xxx 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 h@xxx.xxx.xxx.xxx
        -- Call token is ip$localhost/28088
        -- Call reference is 28088
    -- Called h@xxx.xxx.xxx.xxx
        -- ClearCall: Request to clear call with token
ip$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.
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Krystian.Filiks

2004-Aug-12 06:35 UTC

head link

[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) -------------? Asterisk 
> -----------? H323 GK ------------? 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",
"H323/0797617729@xxx.xxx.xxx.xxx") in
>
>  new stack
>
> Allowed Codecs:
>
>          Table:
>
>    G.723.1{sw} <1>
>
>  Set:
>
>    0:
>
>      0:
>
>        G.723.1{sw} <1>
>
>  
>
>  -- Making call to 0797617729@xxx.xxx.xxx.xxx using gatekeeper.
>
>         == New H.323 Connection created.
>
>         --sj1 is calling host 0797617729@xxx.xxx.xxx.xxx
>
>         -- Call token is ip$localhost/28087
>
>         -- Call reference is 28087
>
>     -- Called 0797617729@xxx.xxx.xxx.xxx
>
>   1:56.153          H225 Caller:813bbe0    h323trans.cxx(656)   
> Trans   Timeout
>
> on request seqnum=14213, try #1 of 2
>
>   1:59.163          H225 Caller:813bbe0    h323trans.cxx(656)   
> Trans   Timeout
>
> on request seqnum=14213, try #2 of 2
>
>         -- Sending SETUP message
>
>         -- Received Facility message...
>
>         -- Received Facility message...
>
>         -- Received Facility message...
>
>         -- Received Facility message...
>
>         -- Received Facility message...
>
>         =*= In CreateRealTimeLogicalChannel for call 28087
>
>                --externalIpAddress: xxx.xxx.xxx.xxx
>
>                --externalPort: 15702
>
>                --SessionID: 1
>
>                -- Direction: IsReceiver
>
>          -- Started logical channel: receiving G.723.1{sw}
>
>                --channelsOpen = 1
>
>         -- Ringing phone for "xxx.xxx.xxx.xxx"
>
>     -- H323/xxx.xxx.xxx.xxx is ringing
>
>   2:10.228          H225 Caller:813bbe0         h323.cxx(2898)  
> H225    Received
>
>  connect PDU.
>
>         =*= In CreateRealTimeLogicalChannel for call 28087
>
>                --externalIpAddress: xxx.xxx.xxx.xxx
>
>                --externalPort: 15702
>
>                --SessionID: 1
>
>                -- Direction: IsTransmitter
>
>          -- Started logical channel: sending G.723.1{sw}
>
>                --channelsOpen = 2
>
>         -- Connection Established with "Tenor Gateway
[xxx.xxx.xxx.xxx]"
>
>     -- H323/xxx.xxx.xxx.xxx answered SIP/sj1-6a47
>
>         -- Received Facility message...
>
>         --ClearCall: Request to clear call with token ip$localhost/28087
>
>         -- Sending RELEASE COMPLETE
>
>   == Spawn extension (default, 0797617729, 1) exited non-zero on 
> 'SIP/sj1-6a47'
>
>     -- Executing Dial("SIP/sj1-6a47",
"H323/h@xxx.xxx.xxx.xxx") in new
> stack
>
> Allowed Codecs:
>
>          Table:
>
>    G.723.1{sw} <1>
>
>  Set:
>
>    0:
>
>      0:
>
>        G.723.1{sw} <1>
>
>  
>
>                channelsOpen = 1
>
>  -- Making call to h@xxx.xxx.xxx.xxx 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 h@xxx.xxx.xxx.xxx
>
>         -- Call token is ip$localhost/28088
>
>         -- Call reference is 28088
>
>     -- Called h@xxx.xxx.xxx.xxx
>
>         --ClearCall: Request to clear call with token ip$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.
>-------------- next part --------------
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administrator tootai

2004-Aug-12 09:01 UTC

head link

[Asterisk-Users] H323 call dropped when answered

Krystian.Filiks a ?crit :
> 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

Krystian Filiks

2004-Aug-12 10:45 UTC

head link

[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: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/h@195.216.65.215")
in new stack
Allowed Codecs:
         Table:
   G.711-uLaw-64k{sw} <1>
 Set:
   0:
     0:
       G.711-uLaw-64k{sw} <1>

 -- Making call to h@195.216.65.215 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 h@195.216.65.215
        -- Call token is ip$localhost/31767
        -- Call reference is 31767
    -- Called h@195.216.65.215
        -- ClearCall: Request to clear call with token ip$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.

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of administrator tootai
Sent: Thursday, August 12, 2004 6:02 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] H323 call dropped when answered

Krystian.Filiks a ?crit :
> 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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Krystian Filiks

2004-Aug-12 13:02 UTC

head link

[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

Krystian.Filiks

2004-Aug-13 03:30 UTC

head link

[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" <Krystian.Filiks@kfiliks.com> 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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>_______________________________________________
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>Asterisk-Users@lists.digium.com
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Krystian.Filiks

2004-Aug-13 03:44 UTC

head link

[Asterisk-Users] H323 call dropped when answered

aaaaaaaaaaaaaaa............
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.
>
>  
>-------------- next part --------------
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Asterisk .

2004-Aug-13 06:56 UTC

head link

[Asterisk-Users] H323 call dropped when answered

--- "Krystian.Filiks" <Krystian.Filiks@kfiliks.com>
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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