Re: [asterisk-users] Asterisk H323 channel and the UDP/TCP rage ports (Q931, H245, T120, RTP)

[Philipp Kempgen]

bilal ghayyad schrieb:

> To be able run Asterisk and gnugk on the same machine at same IP address, I 
> need to know how to configure the port ranges of the (Q931, H245, T120, RTP) 
> for the asterisk H323 channel to avoid any confilict with the gnugk? From 
> where to determine these ranges?
> 
> About gnugk, I know from where to determine it, but I do not know how to 
> determine these port ranges in the Asterisk H323.

Not really an answer to your question but why not simply use
different IP addresses? (bindaddr)


Philipp Kempgen
-- 
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[asterisk-users] Asterisk H323 channel and the UDP/TCP rage ports (Q931, H245, T120, RTP)

[bilal ghayyad]

Hi All;

I am wondering of this H323 channel in asterisk, whatever I ask, I do not get 
help :) - So, how to get help, I do not know.

To be able run Asterisk and gnugk on the same machine at same IP address, I 
need to know how to configure the port ranges of the (Q931, H245, T120, RTP) 
for the asterisk H323 channel to avoid any confilict with the gnugk? From where 
to determine these ranges?

About gnugk, I know from where to determine it, but I do not know how to 
determine these port ranges in the Asterisk H323.

Any help?
Regards
Bilal


  

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[asterisk-users] Asterisk h323 module

[Mikhail Zhirnov]

Hello!

I have a problem with build astersik-addons-1.4.7 on Solaris 10. When I
tried to do "make" I got such error:
*
chan_ooh323.c: In function `reload_config':
chan_ooh323.c:2053: error: `IPTOS_MINCOST' undeclared (first use in this
function)
chan_ooh323.c:2053: error: (Each undeclared identifier is reported only once
chan_ooh323.c:2053: error: for each function it appears in.)
make[1]: *** [chan_ooh323.o] Error 1
make: *** [channels] Error 2*

So I tried to comple PWLIB v.1.10 and haven't success too - the compile
error is :
*
g++ -DP_USE_PRAGMA -D_REENTRANT -Wall  -I/root/pwlib/include -xO3
-DSOLARIS -felide-constructors -c ../../ptclib/pldap.cxx -o
/root/pwlib/lib/obj_solaris_sparc_r/pldap.o
g++: language O3 not recognized
g++: language O3 not recognized
g++: ../common/notifier_ext.cxx: linker input file unused because
linking not done
cc -DSOLARIS -DP_USE_PRAGMA -D_REENTRANT -Wall  -I/root/pwlib/include -c
../common/getdate.tab.c -o /root/pwlib/lib/obj_solaris_sparc_r/getdate.tab.o
make[2]: cc: Command not found
make[2]: *** [/root/pwlib/lib/obj_solaris_sparc_r/getdate.tab.o] Error 127
make[2]: Leaving directory `/root/pwlib/src/ptlib/unix'
make[1]: *** [opt] Error 2
make[1]: Leaving directory `/root/pwlib'
make: *** [optshared] Error 2*


Could you help me to solve the problem?

Thank you,
Mikhail Zhirnov


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Re: [asterisk-users] Asterisk h323 module

[Guillermo Salas M.]

El vie, 05-12-2008 a las 19:04 +0300, Mikhail Zhirnov escribió:
> make[2]: cc: Command not found


Looks like you need cc installed.

Best regards,


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[asterisk-users] Asterisk h323 module

[Mikhail Zhirnov]

Hello!

I have a problem with build astersik-addons-1.4.7 on Solaris 10. When I 
tried to do "make" I got such error:
*
chan_ooh323.c: In function `reload_config':
chan_ooh323.c:2053: error: `IPTOS_MINCOST' undeclared (first use in this 
function)
chan_ooh323.c:2053: error: (Each undeclared identifier is reported only once
chan_ooh323.c:2053: error: for each function it appears in.)
make[1]: *** [chan_ooh323.o] Error 1
make: *** [channels] Error 2*

So I tried to comple PWLIB v.1.10 and haven't success too - the compile 
error is :
*
g++ -DP_USE_PRAGMA -D_REENTRANT -Wall  -I/root/pwlib/include -xO3  
-DSOLARIS -felide-constructors -c ../../ptclib/pldap.cxx -o 
/root/pwlib/lib/obj_solaris_sparc_r/pldap.o
g++: language O3 not recognized
g++: language O3 not recognized
g++: ../common/notifier_ext.cxx: linker input file unused because 
linking not done
cc -DSOLARIS -DP_USE_PRAGMA -D_REENTRANT -Wall  -I/root/pwlib/include -c 
../common/getdate.tab.c -o /root/pwlib/lib/obj_solaris_sparc_r/getdate.tab.o
make[2]: cc: Command not found
make[2]: *** [/root/pwlib/lib/obj_solaris_sparc_r/getdate.tab.o] Error 127
make[2]: Leaving directory `/root/pwlib/src/ptlib/unix'
make[1]: *** [opt] Error 2
make[1]: Leaving directory `/root/pwlib'
make: *** [optshared] Error 2*


Could you help me to solve the problem?

Thank you,
Mikhail Zhirnov

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Re: [asterisk-users] Asterisk H323 Config

[Dovid B]

The same as you would treat any other channel. Specify the default context in 
ooh323.conf (my personal favorite h323 "driver") and in extensions.conf under 
that context set where you want the call to go.
  - Original Message - 
  From: Arun Kumar 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Sunday, October 21, 2007 9:20 AM
  Subject: [asterisk-users] Asterisk H323 Config


  Hi

  Need help on this setup:

  Incoming DID in H323  > Asterisk Server --> SIP Phone


  please tell me to achieve this above setup what needs to be done in Asterisk.


  thanks 

  Arun



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[asterisk-users] Asterisk H323 Config

[Arun Kumar]

Hi

Need help on this setup:

Incoming DID in H323  > Asterisk Server --> SIP Phone


please tell me to achieve this above setup what needs to be done in
Asterisk.


thanks

Arun
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[asterisk-users] Asterisk H323 and Alcatel 4400

[Carlos Chavez]

I have been trying to get an Asterisk 1.2.10 server using ooh323 from
asterisk-addons to call an Alcatel 4400 pbx.  Apart from some stability
problems caused by ooh323 I have not been able to establish a call.
Does anyone have experience with this kind of setup?  Is ooh323 stable
enough for production use?

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Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


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[asterisk-users] Asterisk H323 and Alcatel 4400

[Carlos Chavez]

 Is it possible to talk to an Alcatel 4400 PBX from Asterisk using H323? 
If so, which implementation of h323 would you recommend to do it?  

 So far I have tried with the ooh323 that comes with asterisk-addons but I
cannot seem to establish a call.  If I dial the Alcatel IP I just get a
message that "everyone is busy".  If they dial from the Alcatel I can get a
sip phone to ring but when I answer the call is dropped and I get this message:

Jul 17 17:12:53 WARNING[17865]: src/chan_h323.c:951 ooh323_indicate: Don't
know how to indicate condition -1 on ooh323c_4

 Has anyone tried to get this to work?

--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001

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[Asterisk-Users] Asterisk h323

[Khaled Chehab]

> Hi
>
> How Can asterisk work as sip and h323 protocol in the same time ,and how
is  
> the
> conversion protocol works .
>
> Please if u know send me how to active h323 protocol or the conversion
> protocol
>



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Re[2]: [Asterisk-Users] Asterisk h323

[Grigoriy Puzankin]

On my Asterisk server it (chan_h323) gets 2-3 deadlocks every hour
regardless of openh323/pwlib and asterisk versions (since the
channel_h323 was not updated for a long time). The load is about 25-30
simultaneous calls (from h323 to zaptel, IAX and SIP).

I have another Asterisk server. There's about 5-7 simultaneous calls,
and deadlocks don't occur (calls go from zaptel to h323).

AS> Im using several Asterisk Box with chanh323 from asterisk, and it works
AS> fine.

AS> Sometime it gets deadlocks , but on 1.2.9.1 and 1.2.7 i have estability.
AS> A fail (crash) last month with about 600 calls per day.

AS> Regards

AS> Alberto Sagredo


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Re: [Asterisk-Users] Asterisk h323

[Cesc]

Hi,

I am also testing asterisk with H323, with the channel included in the
latest sources. It works ( i had some problems with media
configuration when calling from an SJPhone ... but it seems more an
SJPhone problem than asterisk).
I also bridged from SIP to H323 ... it works fine.

I have a question for those out there ... i compiled with pwlib 1.9.2
and openh323 1.17.3. These are the versions mentioned in the README
file for the h323 channel.
Now, has anyone tried any newer version? I would be more comfortable
using the latest stable release for these required libraries with
asterisk ... just to make sure i get the best. Any experiences??

Cesc

On 6/20/06, Alberto Sagredo <[EMAIL PROTECTED]> wrote:

Im using several Asterisk Box with chanh323 from asterisk, and it works
fine.

Sometime it gets deadlocks , but on 1.2.9.1 and 1.2.7 i have estability.
A fail (crash) last month with about 600 calls per day.

Regards

Alberto Sagredo


hakem voip escribió:
> You can do this by installing a h323 module.
>
> Conversion works simetimes good, sometimes not good. H323 behaviour on
> asterosk with my experience with kind of unpredictable.
>
>
> 2006/6/20, Khaled Chehab <[EMAIL PROTECTED]
> >:
>
> Hi
>
> Can asterisk work as sip and h323 protocol in the same time ,and
> how is the conversion protocol works .
>
> Please if u know send me how to active h323 protocol or the
> conversion protocol
>
>
>
>
>
>
>
> Regards
>
>
>
> 
> *
> No employee or agent is authorized to conclude any binding
> agreement on behalf of Xplorium with another party by e-mail
> without express written confirmation by an officer of Xplorium.
> Any views expressed by an individual in this electronic message do
> not necessarily reflect views of Xplorium or its subsidiaries and
> associates.
>
> This electronic message and its attachments are solely addressed
> to the addressee(s), and contain confidential information
> protected from disclosure belonging to Xplorium.
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> If you are not the intended addressee of this electronic message
> and its attachments, kindly delete it immediately from your system
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>
>
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Re: [Asterisk-Users] Asterisk h323

[Alberto Sagredo]

Im using several Asterisk Box with chanh323 from asterisk, and it works 
fine.


Sometime it gets deadlocks , but on 1.2.9.1 and 1.2.7 i have estability. 
A fail (crash) last month with about 600 calls per day.


Regards

Alberto Sagredo


hakem voip escribió:

You can do this by installing a h323 module.
 
Conversion works simetimes good, sometimes not good. H323 behaviour on 
asterosk with my experience with kind of unpredictable.


 
2006/6/20, Khaled Chehab <[EMAIL PROTECTED] 
>:


Hi

Can asterisk work as sip and h323 protocol in the same time ,and
how is the conversion protocol works .

Please if u know send me how to active h323 protocol or the
conversion protocol

 

 

 


Regards




*
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Re: [Asterisk-Users] Asterisk h323

[hakem voip]

You can do this by installing a h323 module.
 
Conversion works simetimes good, sometimes not good. H323 behaviour on asterosk with my experience with kind of unpredictable. 
2006/6/20, Khaled Chehab <[EMAIL PROTECTED]>:




Hi 
Can asterisk work as sip and h323 protocol in the same time ,and how is the conversion protocol works .
Please if u know send me how to active h323 protocol or the conversion protocol 
 
 
 
Regards 

*No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects.*  
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Re: [Asterisk-Users] Asterisk h323

[Lenz]

This should provide you enough information to get started.
http://www.astrecipes.net/index.php?q=astrecipes/compiling+asterisk+with+oh323

of course * can operate both SIP and h323 channels, but the support for  
h323 (and I'd add, stability) is not the same you can expect with SIP or  
IAX.

l.


On Tue, 20 Jun 2006 14:23:05 +0200, Khaled Chehab <[EMAIL PROTECTED]>  
wrote:



Hi

Can asterisk work as sip and h323 protocol in the same time ,and how is  
the

conversion protocol works .

Please if u know send me how to active h323 protocol or the conversion
protocol







--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it

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[Asterisk-Users] Asterisk h323

[Khaled Chehab]

Hi 

Can asterisk work as sip and h323 protocol in the same time
,and how is the conversion protocol works .

Please if u know send me how to active h323 protocol or the
conversion protocol 

 

 

 

Regards 






*
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This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.

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Re: [Asterisk-Users] asterisk h323

[leonimar cape]

Hi Yusuf,

I need your info about the installation of the oh323,
I have a problem compiling the pwlib. 

this is the error that I got.. Any ideas,

./configure
checking for g++... no
checking for c++... no
checking for gpp... no
checking for aCC... no
checking for CC... no
checking for cxx... no
checking for cc++... no
checking for cl... no
checking for FCC... no
checking for KCC... no
checking for RCC... no
checking for xlC_r... no
checking for xlC... no
checking for C++ compiler default output file name...
configure: error: C++ compiler cannot create
executables
See `config.log' for more details.


thnx

--- yusuf <[EMAIL PROTECTED]> wrote:

> leonimar cape wrote:
> > Hi,
> > 
> > I just want to inquire which of the available h323
> > modules for asterisk is more stable and better
> > quality. My boss asked me to setup asterisk with
> and I
> > am having a hard time choosing which one should I
> > used. 
> > 
> > Any advice and suggestion will be greatly
> appreciated
> > 
> I have used 0h323 from inAccess Networks,  you
> compile first h323 then 
> asterisk-oh323.  you must use the correct version
> though, read the 
> README.  It worked well for me.  has support for all
> codecs
> 
> I have also used the new ooh323 that comes bundled
> witj asterisk-addons.
> You go into asterisk-addons directoryu and compoile
> ooh323.  Also has 
> worked well.  But this one, as of 1.2.4, only has
> support for ulaw and 
> alaw and gsm i think.  CVS (or SVN) has now support
> for g72*
> 
> both are configured very similiar in oh323.conf and
> ooh323.conf respectivley
> 
> yusuf
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Re: [Asterisk-Users] asterisk h323

[yusuf]

Leonimar,

I cant tell from the error, but i know oh323 is picky about exact 
versions being used.  So in my case i had Asterisk CVS 19/07/2005, i 
used openh323-v1_13_5-src.tar.gz, pwlib-v1_6_6-src.tar.gz, 
asterisk-oh323-0.7.2-pre1.tar.gz


 the README says:

 o PWlib (Portable Text and GUI C/C++ Class Library)
download from http://sourceforge.net/projects/openh323 (v1.6.6)
(required)

  o OpenH323 (Class Library implementing the H.323 protocol)
download from http://sourceforge.net/projects/openh323 (v1.13.5)
(required)

so if you using stable1.2, you must use the correct version of the above


also what does config.log say?

leonimar cape wrote:

Hi Yusuf,

I need your info about the installation of the oh323,
I have a problem compiling the pwlib. 


this is the error that I got.. Any ideas,

./configure
checking for g++... no
checking for c++... no
checking for gpp... no
checking for aCC... no
checking for CC... no
checking for cxx... no
checking for cc++... no
checking for cl... no
checking for FCC... no
checking for KCC... no
checking for RCC... no
checking for xlC_r... no
checking for xlC... no
checking for C++ compiler default output file name...
configure: error: C++ compiler cannot create
executables
See `config.log' for more details.


thnx

--- yusuf <[EMAIL PROTECTED]> wrote:



leonimar cape wrote:


Hi,

I just want to inquire which of the available h323
modules for asterisk is more stable and better
quality. My boss asked me to setup asterisk with


and I


am having a hard time choosing which one should I
used. 


Any advice and suggestion will be greatly


appreciated

I have used 0h323 from inAccess Networks,  you
compile first h323 then 
asterisk-oh323.  you must use the correct version
though, read the 
README.  It worked well for me.  has support for all

codecs

I have also used the new ooh323 that comes bundled
witj asterisk-addons.
You go into asterisk-addons directoryu and compoile
ooh323.  Also has 
worked well.  But this one, as of 1.2.4, only has
support for ulaw and 
alaw and gsm i think.  CVS (or SVN) has now support

for g72*

both are configured very similiar in oh323.conf and
ooh323.conf respectivley

yusuf

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Re: [Asterisk-Users] asterisk h323

[leonimar cape]

Thanks for the info yusuf... Im gonna check it out...

Cheers!

--- yusuf <[EMAIL PROTECTED]> wrote:

> leonimar cape wrote:
> > Hi,
> > 
> > I just want to inquire which of the available h323
> > modules for asterisk is more stable and better
> > quality. My boss asked me to setup asterisk with
> and I
> > am having a hard time choosing which one should I
> > used. 
> > 
> > Any advice and suggestion will be greatly
> appreciated
> > 
> I have used 0h323 from inAccess Networks,  you
> compile first h323 then 
> asterisk-oh323.  you must use the correct version
> though, read the 
> README.  It worked well for me.  has support for all
> codecs
> 
> I have also used the new ooh323 that comes bundled
> witj asterisk-addons.
> You go into asterisk-addons directoryu and compoile
> ooh323.  Also has 
> worked well.  But this one, as of 1.2.4, only has
> support for ulaw and 
> alaw and gsm i think.  CVS (or SVN) has now support
> for g72*
> 
> both are configured very similiar in oh323.conf and
> ooh323.conf respectivley
> 
> yusuf
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>   
>
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> 


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Re: [Asterisk-Users] asterisk h323

[yusuf]

leonimar cape wrote:

Hi,

I just want to inquire which of the available h323
modules for asterisk is more stable and better
quality. My boss asked me to setup asterisk with and I
am having a hard time choosing which one should I
used. 


Any advice and suggestion will be greatly appreciated

I have used 0h323 from inAccess Networks,  you compile first h323 then 
asterisk-oh323.  you must use the correct version though, read the 
README.  It worked well for me.  has support for all codecs


I have also used the new ooh323 that comes bundled witj asterisk-addons.
You go into asterisk-addons directoryu and compoile ooh323.  Also has 
worked well.  But this one, as of 1.2.4, only has support for ulaw and 
alaw and gsm i think.  CVS (or SVN) has now support for g72*


both are configured very similiar in oh323.conf and ooh323.conf respectivley

yusuf
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[Asterisk-Users] asterisk h323

[leonimar cape]

Hi,

I just want to inquire which of the available h323
modules for asterisk is more stable and better
quality. My boss asked me to setup asterisk with and I
am having a hard time choosing which one should I
used. 

Any advice and suggestion will be greatly appreciated

Thanks in advance!

Leonimar

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Re: [Asterisk-Users] asterisk + H323 + 723

[Alberto Sagredo]

Hi, I had the same troubles too.

It does not recognise correctly g723 with oh323. With h323 i have dtmf 
rfc2833 issues but g723 and 729 are transported correctly via H323 
capabilities.


So, let make a try with h323 included in asterisk branch, not the oh323

Kanishka Somaratne wrote:


Hi
I am using asterisk 1.2.1, does any one has any luck with asterisk and 
h323.

I want to use the codecs 723 and 729 with it.
I am having one way audio issues with oh323 with I receive a call to
asterieks through 723 .

is there a successful implementation ?

regards
kani
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[Asterisk-Users] asterisk + H323 + 723

[Kanishka Somaratne]

Hi
I am using asterisk 1.2.1, does any one has any luck with asterisk and h323.
I want to use the codecs 723 and 729 with it.
I am having one way audio issues with oh323 with I receive a call to
asterieks through 723 .

is there a successful implementation ?

regards
kani 


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[Asterisk-Users] asterisk + H323 + 723

[Kanishka Somaratne]

Hi
I am using asterisk 1.2.1, does any one has any luck with asterisk and h323. 
I want to use the codecs 723 and 729 with it.
I am having one way audio issues with oh323 with I receive a call to 
asterieks through 723 .


is there a successful implementation ?

regards
kani 


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Re: [Asterisk-Users] Asterisk H323 Trunk Zone

[Michael Manousos]

Mahmoud Badran wrote:
AVE!
i am trying to register h323 asterisk to the gatekeeper as i installed
asterisk, libpri, zaptel from CVS, and pwlib, openh323, asterisk-oh323
on fedora core3 on a cisco mcs 7800 server problem is i want the
asterisk to register with gatekeeper endpoint with specific zone name
and type...
i searched the web, mail list but there weren't any helpful ones 

could anyone plz tell me how to specify the zone name and type??
You can specify the gatekeeper to use by the zone name using the
following in oh323.conf:
gatekeeper=GKID:
e.g. if the zone name is "MyInternalZone"
gatekeeper=GKID:MyInternalZone
I'm not sure about the type.
Michael.
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[Asterisk-Users] Asterisk H323 Trunk Zone

[Mahmoud Badran]

AVE!

i am trying to register h323 asterisk to the gatekeeper as i installed
asterisk, libpri, zaptel from CVS, and pwlib, openh323, asterisk-oh323
on fedora core3 on a cisco mcs 7800 server problem is i want the
asterisk to register with gatekeeper endpoint with specific zone name
and type...

i searched the web, mail list but there weren't any helpful ones 

could anyone plz tell me how to specify the zone name and type??

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[Asterisk-Users] Asterisk-h323

[gale81]

Hi
I've this plattaform

sip sjphone - - - asterisk- - -gatekeeper- - -ohphone- - -phonejack card-
- analog phone

Asterisk is registered with Gatekeeper
Ohphone is registered with Gatekeeper
Phonejack is installed successfully and get a dial tone

When i try to call phonejack with sip phone i have this message :
call with mysjphone ('alias/IP' of asterisk) completed duration 0:00
and analog phone don't ring

when i try to call mysjphone with analog phone i've this message:

Speed Dial 3231 not defined ,trying gatekeeper..
phonejack is calling host 3231
'alias'   'ip' of asterisk is busy duration 0:01

Have you suggestions?
Thanks Ale   

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[Asterisk-Users] Asterisk, h323

[Osman İZBAT]

Hello
I've installed asterisk with [EMAIL PROTECTED] package with h323 support.
I've a Digium TDM10B card and we have a quintum voip gateway. I'm
trying to make call with an analog phone plugged to that card through
our quintum with h323 protocol.
How to confgure related files? Any help welcome.

Thanks.
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[Asterisk-Users] asterisk-h323 and h323_id

[Sam Njenga]

Hi all
 
Has anyone managed to send an outgoing call using 
asterisk-h323 and successfully sent the H323_id ?
 
Sam
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Re: [Asterisk-Users] Asterisk H323 support

[Nardis Dome]

> Date: Mon, 21 Feb 2005 00:20:39 -0800 (PST)
> [zone:-], [EMAIL PROTECTED]
> mentioned in msg:  H323 support> that ...
> 
> > with Openh323 - v1.12.2 and pwlib - v1.5.2 I use
> > asterisk-oh323 v.0.6.3b and it works fine
> 
> What version of Asterisk are you running? And on
> what os and distribution?
> 
> --
> Kuniyoshi
>
Murata.iChat/AIM:macwebcaster
> English-Japanese Interpreter
> mailto:[EMAIL PROTECTED]
> Macintosh Webcast Specialist   
> http://www.macwebcaster.com
> 

Hi,

Asterisk 1.0.1
CentOS 3.3 24.21-20.EL.c0
pwlib 1.5.2
openh323 1.12.2
asterisk-oh323 0.6.3b



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Re: [Asterisk-Users] Asterisk H323 support

[Kuniyoshi Murata]

Hi,
(B
(BI'm using Asterisk-1.0.0 on Fedora Core 1
(B
(BDate: Mon, 21 Feb 2005 14:22:13 +0900 [zone:Tokyo],
(B[EMAIL PROTECTED] mentioned in msg:  that ...
(B
(B>   For channel asterisk-oh323-v0.6.5
(B>   need
(B>   openh323-Janus_patch4-src-tar.gz
(B>   pwlib-Janus_patch4-src-tar.gz
(B> 
(B
(BI tried this combination but openh323 fails in compiling (make clean, make
(Bopt).
(B
(BDate: Mon, 21 Feb 2005 00:20:39 -0800 (PST) [zone:-], [EMAIL PROTECTED]
(Bmentioned in msg:  that ...
(B
(B> with Openh323 - v1.12.2 and pwlib - v1.5.2 I use
(B> asterisk-oh323 v.0.6.3b and it works fine
(B
(BWhat version of Asterisk are you running? And on what os and distribution?
(B
(B--
(BKuniyoshi Murata.iChat/AIM:macwebcaster
(BEnglish-Japanese Interpreter mailto:[EMAIL PROTECTED]
(BMacintosh Webcast Specialisthttp://www.macwebcaster.com
(B
(B
(B
(B___
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(BAsterisk-Users@lists.digium.com
(Bhttp://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Asterisk H323 support

[Nardis Dome]

Hi,

with Openh323 - v1.12.2 and pwlib - v1.5.2 I use
asterisk-oh323 v.0.6.3b and it works fine

hope it helps

cu...



--- kolo sos <[EMAIL PROTECTED]> wrote:

> Hi,
> 
> anybody knows what's missing or problem why i cant
> compile asterisk-oh323 in my machine?
> 
> i got this compiled successfully
> 
> ...Openh323 - v1.12.2
> ...pwlib - v1.5.2
> 
> except 
> 
> ...asterisk-oh323 - v0.6.5
> 
> here's the output as i run make...
> 
> [EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$ make
> for x in wrapper asterisk-driver; do make -C $x
> build
> || exit 1 ; done
> make[1]: Entering directory
> `/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper'
> ./check_ver /home/mkoy/pwlib pwlib
> ./check_ver /home/mkoy/openh323 openh323
> g++ -DP_LINUX=2.4.26 -ffunction-sections
> -fdata-sections -D_REENTRANT -Wall -fPIC
> -DP_USE_PRAGMA -DPHAS_TEMPLATES
> -I/home/mkoy/pwlib/include/ptlib/unix
> -I/usr/include/pwlib -I/home/mkoy/pwlib/include
> -DPTRACING -I/home/mkoy/openh323/include -DHAS_IXJ
> -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING
> -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.5.2\"
> -DOPENH323VERSION=\"1.12.2\" 
> -I/home/mkoy/pwlib/include/ptlib/unix
> -I/home/mkoy/pwlib/include
> -I/home/mkoy/openh323/include
> -I/home/mkoy/openh323/include/openh323
> -I../asterisk-driver -c asteriskaudio.cxx -o
> asteriskaudio.o
> asteriskaudio.cxx: In destructor `virtual
>PAsteriskSoundChannel::~PAsteriskSoundChannel()':
> asteriskaudio.cxx:167: error: `baseChannel'
> undeclared
> (first use this
>function)
> asteriskaudio.cxx:167: error: (Each undeclared
> identifier is reported only once
>for each function it appears in.)
> make[1]: *** [asteriskaudio.o] Error 1
> make[1]: Leaving directory
> `/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper'
> make: *** [subdirs_build] Error 1
> [EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$
> 
> 
> 
> Kolosos
> Philippines
> 
> 
>   
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>   
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Re: [Asterisk-Users] Asterisk H323 support

[Андрей Кочетков]

Здравствуйте Asterisk Users Mailing List - Non-Commercial Discussion,

Monday, February 21, 2005, 2:14:20 PM, Вы писали:

===8<==Original message text===
kolo sos> Hi,

kolo sos> anybody knows what's missing or problem why i cant
kolo sos> compile asterisk-oh323 in my machine?

kolo sos> i got this compiled successfully

kolo sos> Openh323 - v1.12.2
kolo sos> pwlib - v1.5.2

kolo sos> except 

kolo sos> asterisk-oh323 - v0.6.5

kolo sos> here's the output as i run make...

kolo sos> [EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$ make
kolo sos> for x in wrapper asterisk-driver; do make -C $x build
kolo sos> || exit 1 ; done
kolo sos> make[1]: Entering directory
kolo sos> `/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper'
kolo sos> ../check_ver /home/mkoy/pwlib pwlib
kolo sos> ../check_ver /home/mkoy/openh323 openh323
kolo sos> g++ -DP_LINUX=2.4.26 -ffunction-sections
kolo sos> -fdata-sections -D_REENTRANT -Wall -fPIC
kolo sos> -DP_USE_PRAGMA -DPHAS_TEMPLATES
kolo sos> -I/home/mkoy/pwlib/include/ptlib/unix
kolo sos> -I/usr/include/pwlib -I/home/mkoy/pwlib/include
kolo sos> -DPTRACING -I/home/mkoy/openh323/include -DHAS_IXJ
kolo sos> -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING
kolo sos> -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.5.2\"
kolo sos> -DOPENH323VERSION=\"1.12.2\" 
kolo sos> -I/home/mkoy/pwlib/include/ptlib/unix
kolo sos> -I/home/mkoy/pwlib/include
kolo sos> -I/home/mkoy/openh323/include
kolo sos> -I/home/mkoy/openh323/include/openh323
kolo sos> -I../asterisk-driver -c asteriskaudio.cxx -o
kolo sos> asteriskaudio.o
kolo sos> asteriskaudio.cxx: In destructor `virtual
kolo sos>PAsteriskSoundChannel::~PAsteriskSoundChannel()':
kolo sos> asteriskaudio.cxx:167: error: `baseChannel' undeclared
kolo sos> (first use this
kolo sos>function)
kolo sos> asteriskaudio.cxx:167: error: (Each undeclared
kolo sos> identifier is reported only once
kolo sos>for each function it appears in.)
kolo sos> make[1]: *** [asteriskaudio.o] Error 1
kolo sos> make[1]: Leaving directory
kolo sos> `/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper'
kolo sos> make: *** [subdirs_build] Error 1
kolo sos> [EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$



kolo sos> Kolosos
kolo sos> Philippines



kolo sos> __ 
kolo sos> Do you Yahoo!? 
kolo sos> Meet the all-new My Yahoo! - Try it today! 
kolo sos> http://my.yahoo.com 
 

kolo sos> ___
kolo sos> Asterisk-Users mailing list
kolo sos> Asterisk-Users@lists.digium.com
kolo sos> http://lists.digium.com/mailman/listinfo/asterisk-users
kolo sos> To UNSUBSCRIBE or update options visit:
kolo sos>http://lists.digium.com/mailman/listinfo/asterisk-users

===8<===End of original message text===

  For channel asterisk-oh323-v0.6.5
  need
  openh323-Janus_patch4-src-tar.gz
  pwlib-Janus_patch4-src-tar.gz


-- 
С уважением:
Андрей Кочетков
ООО "Современные Системы Связи", "Мобил-Телеком"
Чита, ул. Заб. Рабочего, 94
тел.: 8 (3022) 23-33-33 
mailto:[EMAIL PROTECTED]

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[Asterisk-Users] Asterisk H323 support

[kolo sos]

Hi,

anybody knows what's missing or problem why i cant
compile asterisk-oh323 in my machine?

i got this compiled successfully

...Openh323 - v1.12.2
...pwlib - v1.5.2

except 

...asterisk-oh323 - v0.6.5

here's the output as i run make...

[EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$ make
for x in wrapper asterisk-driver; do make -C $x build
|| exit 1 ; done
make[1]: Entering directory
`/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper'
./check_ver /home/mkoy/pwlib pwlib
./check_ver /home/mkoy/openh323 openh323
g++ -DP_LINUX=2.4.26 -ffunction-sections
-fdata-sections -D_REENTRANT -Wall -fPIC
-DP_USE_PRAGMA -DPHAS_TEMPLATES
-I/home/mkoy/pwlib/include/ptlib/unix
-I/usr/include/pwlib -I/home/mkoy/pwlib/include
-DPTRACING -I/home/mkoy/openh323/include -DHAS_IXJ
-DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING
-DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.5.2\"
-DOPENH323VERSION=\"1.12.2\" 
-I/home/mkoy/pwlib/include/ptlib/unix
-I/home/mkoy/pwlib/include
-I/home/mkoy/openh323/include
-I/home/mkoy/openh323/include/openh323
-I../asterisk-driver -c asteriskaudio.cxx -o
asteriskaudio.o
asteriskaudio.cxx: In destructor `virtual
   PAsteriskSoundChannel::~PAsteriskSoundChannel()':
asteriskaudio.cxx:167: error: `baseChannel' undeclared
(first use this
   function)
asteriskaudio.cxx:167: error: (Each undeclared
identifier is reported only once
   for each function it appears in.)
make[1]: *** [asteriskaudio.o] Error 1
make[1]: Leaving directory
`/home/mkoy/voip/asterisk-oh323-0.6.5/wrapper'
make: *** [subdirs_build] Error 1
[EMAIL PROTECTED]:~/voip/asterisk-oh323-0.6.5$



Kolosos
Philippines



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Re: [Asterisk-Users] Asterisk-H323

[Martijn van Oosterhout]

On Thu, Feb 17, 2005 at 10:47:28PM -0800, kolo sos wrote:
> is there any version mismatch or path needed to have a
> succesful build? i got an error when i done MAKE to
> the asterisk-oh323.

Obviously people have successfully built it, people here use it all the
time. Perhaps you can post the actual error you're getting...
-- 
Martijn van Oosterhout
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[Asterisk-Users] Asterisk-H323

[kolo sos]

hi,

im trying to add H323 capability of asterisk to
communicate with my VOIP gateway which is H323
compliant. but it seems no luck for me to have error
free binary build with my machine. 

I downloaded the latest aserisk-oh323-0.7.1 and the
required libraries (Openh323 1.13.5, pwlib 1.6.6).

is there any version mismatch or path needed to have a
succesful build? i got an error when i done MAKE to
the asterisk-oh323.

anybody have tried to a succesfull build...i badly
need your hand on this...

k0l0s0s
Philippines



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RE: [Asterisk-Users] Asterisk-H323

[Vitalie Apostu]

 No, I am using H323 driver

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Liu
Sent: Monday, February 14, 2005 11:36 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk-H323

Hi there,

The settings are in oh323.conf

; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;

I assume you are using the OH323 driver right?

Also if no audio, it could also be a codec issue. You need to set the codec
for the OH323 call in oh323.conf as well.

David
Hong Kong

On Mon, 14 Feb 2005 11:27:53 -0500, Vitalie Apostu wrote
> Cisco and Asterisk are not behind firewall.
> 
> Where can I check for settings noH245Tuneling and noFastStart in 
> Asterisk H323?
> 
> -
> -- Executing Dial("SIP/msn-069a", "H323/[EMAIL PROTECTED]:1720") in 
> new stack-- Called [EMAIL PROTECTED]:1720-- H323/peer:1720 is 
> making progress passing it to SIP/msn-069a-- H323/peer:1720 is ringing
> -- H323/peer:1720 answered SIP/msn-069a
>   == Spawn extension (messanger, 73952389506, 1) exited non-zero on 
> 'SIP/msn-069a'
> --
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Monday, February 14, 2005 11:29 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Asterisk-H323
> 
> Make sure settings for:
> 
> noH245Tuneling and noFastStart parameters are correctly tuned both 
> sides.
> 
> Is Cisco or Asterisk behind NAT?
> 
> Send more info
> 

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RE: [Asterisk-Users] Asterisk-H323

[Vitalie Apostu]

noH245Tunneling instead of noH245Tuneling

 typedef struct call_options {

charcid_num[80];

charcid_name[80];

int noFastStart;

int noH245Tunneling;

int noSilenceSuppression;

unsigned intport;

int progress_setup;

int progress_alert;

int progress_audio;

int dtmfcodec;

} call_options_t;

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Monday, February 14, 2005 11:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk-H323

Make sure settings for: 

noH245Tuneling and noFastStart parameters are correctly tuned both sides. 

Is Cisco or Asterisk behind NAT? 

Send more info



> 
> Greetings,
> 
> I have a problem making a call from Asterisk to Cisco H323 PSTN 
> gateway using H323 channel. I can call but there are no sound in both 
> way. If I call
> H323 gateway directly from SJPhone I have no problem with sound.
> 
> Any advice are welcome.
> 
> Thanks in advance.
> 
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> 

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RE: [Asterisk-Users] Asterisk-H323

[David Liu]

Hi there,

The settings are in oh323.conf

; Enable fast start (yes,no).
;
fastStart=yes
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;

I assume you are using the OH323 driver right?

Also if no audio, it could also be a codec issue. You need to set the codec 
for the OH323 call in oh323.conf as well.

David
Hong Kong

On Mon, 14 Feb 2005 11:27:53 -0500, Vitalie Apostu wrote
> Cisco and Asterisk are not behind firewall.
> 
> Where can I check for settings noH245Tuneling and noFastStart in Asterisk
> H323?
> 
> -
> -- Executing Dial("SIP/msn-069a", "H323/[EMAIL PROTECTED]:1720") in 
> new stack-- Called [EMAIL PROTECTED]:1720-- H323/peer:1720 is 
> making progress passing it to SIP/msn-069a-- H323/peer:1720 is ringing
> -- H323/peer:1720 answered SIP/msn-069a
>   == Spawn extension (messanger, 73952389506, 1) exited non-zero on
> 'SIP/msn-069a'
> --
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
> Sent: Monday, February 14, 2005 11:29 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Asterisk-H323
> 
> Make sure settings for:
> 
> noH245Tuneling and noFastStart parameters are correctly tuned both 
> sides.
> 
> Is Cisco or Asterisk behind NAT?
> 
> Send more info
> 

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RE: [Asterisk-Users] Asterisk-H323

[Vitalie Apostu]

Cisco and Asterisk are not behind firewall.

Where can I check for settings noH245Tuneling and noFastStart in Asterisk
H323?

-
-- Executing Dial("SIP/msn-069a", "H323/[EMAIL PROTECTED]:1720") in new stack
-- Called [EMAIL PROTECTED]:1720
-- H323/peer:1720 is making progress passing it to SIP/msn-069a
-- H323/peer:1720 is ringing
-- H323/peer:1720 answered SIP/msn-069a
  == Spawn extension (messanger, 73952389506, 1) exited non-zero on
'SIP/msn-069a'
--

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Monday, February 14, 2005 11:29 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk-H323

Make sure settings for: 

noH245Tuneling and noFastStart parameters are correctly tuned both sides. 

Is Cisco or Asterisk behind NAT? 

Send more info



> 
> Greetings,
> 
> I have a problem making a call from Asterisk to Cisco H323 PSTN 
> gateway using H323 channel. I can call but there are no sound in both 
> way. If I call
> H323 gateway directly from SJPhone I have no problem with sound.
> 
> Any advice are welcome.
> 
> Thanks in advance.
> 
> ___
> Asterisk-Users mailing list
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> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 

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[Asterisk-Users] Asterisk-H323

[Vitalie Apostu]

Greetings,

I have a problem making a call from Asterisk to Cisco H323 PSTN gateway
using H323 channel. I can call but there are no sound in both way. If I call
H323 gateway directly from SJPhone I have no problem with sound.

Any advice are welcome.

Thanks in advance.

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Re: [Asterisk-Users] asterisk+h323+rh9

[Ginel Tudorache]

Hi,
Thanks for your help. I downloaded "Janus-patched" of pwlib and OpenH323 
from sourceforge.net, asterisk-0.6.5 from 
http://www.inaccessnetworks.com and asterisk 1.0.5 from asterisk.org.
All these were compiled without problems and now I have a running 
asterisk pbx with oh323.
Now i'm trying to learn how to config asterisk :(!

Kind regards,
Ginel
Roger Schreiter wrote:
Ginel Tudorache schrieb:
Hi,
I'm trying to install asterisk with h323 support on rh9 box. I want to 
find a working combination between asterisk,asterisk-oh323,pwlib and 
openh323.

Hi,
I'm using SuSE, not Redhat, but imho you'll succeed, if
you strictly follow the version hints mentioned in
asterisk-oh's README file.
I downloaded and installed asterisk-oh several times last
year. Last time was November:
- asterisk-oh323-0.7.0
- pwlib-1.6.6, I downloaded "Janus-patched" version from
  http://www.inaccessnetworks.com
- openh323-1.13.5, I downloaded "Janus-patched" version from
  http://www.inaccessnetworks.com
  and applied patch mentioned in the asterisk-oh323-0.7.0-README
  file.
Hope that helps!
Roger.
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Re: [Asterisk-Users] asterisk+h323+rh9

[Roger Schreiter]

Ginel Tudorache schrieb:
Hi,
I'm trying to install asterisk with h323 support on rh9 box. I want to 
find a working combination between asterisk,asterisk-oh323,pwlib and 
openh323.

Hi,
I'm using SuSE, not Redhat, but imho you'll succeed, if
you strictly follow the version hints mentioned in
asterisk-oh's README file.
I downloaded and installed asterisk-oh several times last
year. Last time was November:
- asterisk-oh323-0.7.0
- pwlib-1.6.6, I downloaded "Janus-patched" version from
  http://www.inaccessnetworks.com
- openh323-1.13.5, I downloaded "Janus-patched" version from
  http://www.inaccessnetworks.com
  and applied patch mentioned in the asterisk-oh323-0.7.0-README
  file.
Hope that helps!
Roger.
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[Asterisk-Users] asterisk+h323+rh9

[Ginel Tudorache]

Hi,
I'm trying to install asterisk with h323 support on rh9 box. I want to 
find a working combination between asterisk,asterisk-oh323,pwlib and 
openh323.
Thank you!

Ginel Tudorache
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Re: [Asterisk-Users] Asterisk H323 Trunk

[Jeremy McNamara]

H.323 doesn't have an explicit caller*id feature, so any callerid 
specific features that have been added are hacks.  Since you are using a 
gatekeeper why don't you use a type=h323 to specify your H.323 id properly?

[6400047602100]
type=h323
secret=securepassword   ; optional
Find me on IRC (JerJer) if u want to discuss this.

Jeremy McNamara





Roger De Salis wrote:

During debugging of H323 trunk side (using Jeremy Macnamara's H323
driver in ~/channels/h323) a couple of things come don't quite work
as advertised...
1/ the following line in extensions.conf explicitly sets the
   outgoing caller ID (required in my case for downstream GK
   processing..)
exten => _1NX.,1,SetCallerID,6400047602100
exten => _1NX.,2,Dial,H323/${EXTEN:1}
   what comes out in the h323 datastream and logs is:-

 dialled digits
{
{src digits}
{ "6400047602100)" }
...
note the unbalanced closing backet. We tried changing the
number length, and doing called ID with different functions,
but it looks like a bug.
A fairly detailed squiz around the digium site did not point
where to file a bug to, so I apologise for polluting this list..
=

When the h323 channel driver registers..

vipe50#sho gatek endp
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name Type
Flags
--- - --- - - 
-
202.37.19.101720  202.37.19.101024  zone1 TERM
E164-ID: 6400047602201
202.37.19.111720  202.37.19.111719  zone1 TERM
E164-ID: 6400047602999
;
; Asterisk, strongly preferred as a VOIP-GW, but registers as a TERM.
;
202.37.19.121720  202.37.19.1232829 zone1 TERM
H323-ID: fxchange
E164-ID: 6400047602100
;
; Cisco Call Manager, registering as a VOIP GW, with H323 Trunk
;
202.37.83.101720  202.37.83.101710  zone1 VOIP-GW
H323-ID: 202.37.83.10
203.79.85.252   1720  203.79.85.252   1719  zone1 TERM
E164-ID: 6400047601020

I had a look through the source, no comments stood out, any know a way
to get * registered as a VOIP-GW, rather than a TERM? Played with all 
the obvious things in h323.conf

Many Thanks for reading this far...

Rgds Roger De Salis


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[Asterisk-Users] Asterisk H323 Trunk

[Roger De Salis]

During debugging of H323 trunk side (using Jeremy Macnamara's H323
driver in ~/channels/h323) a couple of things come don't quite work
as advertised...
1/ the following line in extensions.conf explicitly sets the
   outgoing caller ID (required in my case for downstream GK
   processing..)
exten => _1NX.,1,SetCallerID,6400047602100
exten => _1NX.,2,Dial,H323/${EXTEN:1}
   what comes out in the h323 datastream and logs is:-

 dialled digits
{
{src digits}
{ "6400047602100)" }
...
note the unbalanced closing backet. We tried changing the
number length, and doing called ID with different functions,
but it looks like a bug.
A fairly detailed squiz around the digium site did not point
where to file a bug to, so I apologise for polluting this list..
=

When the h323 channel driver registers..

vipe50#sho gatek endp
GATEKEEPER ENDPOINT REGISTRATION

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name TypeFlags
--- - --- - - -
202.37.19.101720  202.37.19.101024  zone1 TERM
E164-ID: 6400047602201
202.37.19.111720  202.37.19.111719  zone1 TERM
E164-ID: 6400047602999
;
; Asterisk, strongly preferred as a VOIP-GW, but registers as a TERM.
;
202.37.19.121720  202.37.19.1232829 zone1 TERM
H323-ID: fxchange
E164-ID: 6400047602100
;
; Cisco Call Manager, registering as a VOIP GW, with H323 Trunk
;
202.37.83.101720  202.37.83.101710  zone1 VOIP-GW
H323-ID: 202.37.83.10
203.79.85.252   1720  203.79.85.252   1719  zone1 TERM
E164-ID: 6400047601020
I had a look through the source, no comments stood out, any know a way
to get * registered as a VOIP-GW, rather than a TERM? Played with all 
the obvious things in h323.conf

Many Thanks for reading this far...

Rgds Roger De Salis
--
 \_Roger De Salis   rdesalis at fx dot net dot nz
 
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[Asterisk-Users] Asterisk H323 endpoint (ATA 186)

[Santosh Prasad]

Hello,


In reference to the mailing list as shown below

http://lists.digium.com/pipermail/asterisk-users/2003-February/007371.html

George:

   Hi, if you get the answer can you please share it with me. Thanks...

[EMAIL PROTECTED] said:

> Hi all,
> 
> How can I make asterisk to route all outgoing calls to a H.323 port ?
And all
> incomming call be routed from the H323 to the proper extension ?
what/which
> files should I specify with existing h323 channel drviver ?
> 
> George


I am trying to make  H323 endpoints (ATA 186)  talk to each other. I am
able to place calls successfully with gnugk (gatekeeper). Now I would   
like to know how to configure oh323.conf or any other required file for
these end points to place calls through asterisk. Also I would like to  
add that asterisk is registered to gate keeper as H.323 end point.  

Thanks

Santosh

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[Asterisk-Users] asterisk h323 backtrace

[Makerere University]

i have compiled h323 support into asterisk using channels/h323. Sometimes 
asterisk runs and other times it segments. I have dumped the core using 
asterisk -vvg and this is the back trace any pointers 

Core was generated by `asterisk -vvg'.
Program terminated with signal 11, Segmentation fault.
#0  0x400d30a6 in ?? ()
(gdb) bt
#0  0x400d30a6 in ?? ()
#1  0x400305e0 in ?? ()
#2  0x40030714 in ?? ()
#3  0x406d5abb in ?? ()
#4  0x406d7025 in ?? ()
#5  0x8053b46 in ?? ()
#6  0x8053fa6 in ?? ()
#7  0x807ab1b in ?? ()
#8  0x400c2c6f in ?? ()
(gdb) 
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