Re: [asterisk-users] Looking for Asterisk Consultants & Experts

2015-09-02 Thread Ganbold Tsagaankhuu 
Shahid,

On Wed, Sep 2, 2015 at 8:40 PM, Shahid H  wrote:

> Hello,
>
> Can someone recommend me where is best place to find Asterisk
> Expert/Consultant for freelance work?
>
> If you are interested to work as a freelancer, you can email me directly.
>
>
What kind of work do you have?

thanks,

Ganbold




> Thanks
>
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[asterisk-users] chan_woomera tries to connect to strange host

2008-02-21 Thread Ganbold Tsagaankhuu 
Hi,

It is very strange that following chan_woomera code part gives IP address
44.215.5.41.

static int connect_woomera(int *new_socket, woomera_profile *profile, int
flags)
{
struct sockaddr_in localAddr, remoteAddr;
struct hostent *hp;
struct ast_hostent ahp;
int res = 0;

*new_socket=-1;

printf(WOOMERA HOST: %s\n,profile-woomera_host); // THIS PRINTS CORRECT
IP ADDRESS
if ((hp = ast_gethostbyname(profile-woomera_host, ahp))) {
remoteAddr.sin_family = hp-h_addrtype; memcpy((char *)
remoteAddr.sin_addr.s_addr, hp-h_addr_list[0], hp-h_length);

printf( WOOMERA HOST: %s\n, inet_ntoa(
remoteAddr.sin_addr)); // THIS PRINTS 44.215.5.41
remoteAddr.sin_port = htons(profile-woomera_port);

And chan_woomera tries to connect to 44.215.5.41:42240, it is very strange,
because
I have already defined host 192.168.0.18 in woomera.conf.

I can hardcode it like:

// memcpy((char *) remoteAddr.sin_addr.s_addr, hp-h_addr_list[0],
hp-h_length);
inet_aton(192.168.0.18, remoteAddr.sin_addr);

But I would like to know the right solution.
Please let me know.

thanks,

Ganbold
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[asterisk-users] patch which makes Asterisk-Addons 1.4.5 work when codec negotiation patch applied to asterisk

2008-02-14 Thread Ganbold Tsagaankhuu 
Hi,

Since the original codec negotiation patch (
http://bugs.digium.com/view.php?id=4825 report) just closed yesterday,
and as well as my report (http://bugs.digium.com/view.php?id=11998), I had
nothing to do but send my patches to the list.
It might be good if my patches are placed at
http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch, but
don't know whom should I contact.
Anyway sending here.

thanks,

Ganbold


chan_h323.c.patch1
Description: Binary data


ooh323cDriver.c.patch
Description: Binary data
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Re: [asterisk-users] [asterisk-dev] chan_ooh323 patches compatible with codec negotiation patch applied to asterisk 1.4.17

2008-02-13 Thread Ganbold Tsagaankhuu 
Hi all,

It is posted here:

http://bugs.digium.com/view.php?id=11976

Still waiting for the approval.

Please see the notes.

thanks,

Ganbold


On 2/12/08, Johan Wilfer [EMAIL PROTECTED] wrote:

 Ganbold Tsagaankhuu wrote:
  Hi all,
 
  Sorry for cross posting.
  I attached my chan_ooh323 patches (asterisk-addons-1.4.5) when codec
  negotiation patch changes applied to asterisk-1.4.17.
  Please let me know whether my patches are correct or not.
 
  thanks in advance,
 
  Ganbold
 
 
  
 
  
 For licensing issues nobody will be able to use your patch if you don't
 submit it thought the bug tracker at http://bugs.digium.com/
 You will be able to agree to the digium license after you have created
 an account.
 There is also a bug tracker introduction that is useful to read at
 http://asterisk.org/developers/bug-guidelines

 Nice work!
 /Johan

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[asterisk-users] chan_ooh323 patches compatible with codec negotiation patch applied to asterisk 1.4.17

2008-02-11 Thread Ganbold Tsagaankhuu 
Hi all,

Sorry for cross posting.
I attached my chan_ooh323 patches (asterisk-addons-1.4.5) when codec
negotiation patch changes applied to asterisk-1.4.17.
Please let me know whether my patches are correct or not.

thanks in advance,

Ganbold


chan_h323.c.patch
Description: Binary data


ooh323cDriver.c.patch
Description: Binary data
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[asterisk-users] asterisk, asterisk-addons, ooh323, codec negotiation patch in FreeBSD

2008-02-10 Thread Ganbold Tsagaankhuu 
Hi,

Sorry for cross posting.
I'm trying to use asterisk+ooh323 in FreeBSD 7.0-PRERELEASE.
It seems like asterisk-addons is marked as broken in FreeBSD-7.0.
Can somebody tell me the reason why it is broken if somebody knows?
And also can I try newer versions of it? Basically I need ooh323 with codec
negotiation patch.
Any idea, experience is greatly appreciated.

thanks in advance,

Ganbold
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[Asterisk-Users] chan_h323 problem

2006-03-23 Thread Ganbold Tsagaankhuu 
Hello,

I installed Asterisk from CVS on Redhat Linux 9 and working with
chan_h323 module and g729/g723 free codecs too.

My network connection diagram:
--
X-lite/X-Pro--Asterisk--chan_h323--GnuGK---AS5300--PSTN

boldsoft*CLI show version
Asterisk CVS-v1-0-03/24/06-15:27:08 built by [EMAIL PROTECTED] on a i686
running Linux
I can make H323 call without any problem from X-Pro and from X-lite
dead-air both end.

My default h323.conf
===
[general]
port = 1720
bindaddr = 0.0.0.0

disallow=all
allow=g729
allow=g723
allow=alaw
allow=ulaw
allow=gsm

gatekeeper = a.b.c.d
AllowGKRouted = yes
noFastStart = no
noH245Tunneling = no
noSilenceSuppression = no
Modified h323.conf
==
[general]
port = 1720
bindaddr = 0.0.0.0

disallow=all
allow=g729
allow=g723
allow=alaw
allow=ulaw
allow=gsm

gatekeeper = a.b.c.d
AllowGKRouted = yes
noFastStart = yes
noH245Tunneling = yes
noSilenceSuppression = no
I can to hear one-way audio from X-lite side, but no audio from PSTN side

I did upgrade Asterisk to Asterisk-1.2.5 and chan_h323 doesn't work
default and even modified config.

Any suggestion? Which H323 channel module is better (chan_h323, oh323, ooh323)?

I downloaded ooh323c 0.8.1 and don't know how to create asterisk
module using source?

Regards,
Ganbold
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[Asterisk-Users] Asterisk with GnuGK

2006-01-18 Thread Ganbold Tsagaankhuu 
Hello,I am trying to configure asterisk with GnuGK.Is it possible to configure Asterisk OH323, CHAN_H323 without register to GnuGK gatekeeper?thanks,Ganbold
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[Asterisk-Users] Few questions about Asterisk

2005-08-02 Thread Ganbold Tsagaankhuu 
 kernel: DIAL: T95158330w
Jul 21 02:48:57 boldsoft kernel: DIAL: T95158330w
Jul 21 02:49:20 boldsoft kernel: DIAL: T345598w
Jul 21 02:49:43 boldsoft kernel: DIAL: T322473w
Jul 21 02:50:02 boldsoft kernel: DIAL: T99114909w
Jul 21 02:50:27 boldsoft kernel: DIAL: T99190040w
Jul 21 02:50:48 boldsoft kernel: DIAL: T95158330w
Jul 21 02:51:11 boldsoft kernel: DIAL: T91911187w
Jul 21 02:54:01 boldsoft kernel: DIAL: T95158330w
Jul 21 02:54:33 boldsoft kernel: DIAL: T345598w
Jul 21 03:04:35 boldsoft kernel: DIAL: T345598w
Jul 21 03:06:00 boldsoft kernel: DIAL: T99190040w
Jul 21 03:06:24 boldsoft kernel: DIAL: T11345598w
Jul 21 03:12:11 boldsoft kernel: DIAL: T1109w
Jul 21 03:13:59 boldsoft kernel: DIAL: T18340w
Jul 21 03:14:18 boldsoft kernel: DIAL: T18000w
Jul 21 03:14:31 boldsoft kernel: DIAL: T18300w
Jul 21 03:14:46 boldsoft kernel: DIAL: T345598w
Jul 21 03:15:16 boldsoft kernel: DIAL: T99190040w
Jul 21 03:15:46 boldsoft kernel: DIAL: T686000w
Jul 21 03:16:09 boldsoft kernel: DIAL: T341341w
Jul 21 03:16:39 boldsoft kernel: DIAL: T99114909w
Jul 21 03:17:02 boldsoft kernel: DIAL: T345598w
Jul 21 09:47:15 boldsoft kernel: DIAL: T345598w

In Mongolia PSTN local number has 5-6 digits, Mobile number has 8
digits and special enquiry service with 3-4 digits.

I can to hear BUSY TONE after several seconds.

4.There another problem with Wildcard TDM400P REV E/F FXS port.

After ~30mins I can't to hear DIALTONE on analog phone connected to
FXS port.

and I got following in /var/log/messages
Jul 21 01:35:39 boldsoft kernel: Power alarm on module 1, resetting!
Jul 21 09:57:06 boldsoft kernel: Freed a Wildcard
Jul 21 09:57:06 boldsoft kernel: ZapTel detach!
Jul 21 09:57:06 boldsoft kernel: wcfxo0: detached
Jul 21 09:57:13 boldsoft kernel: FXS device: vendor=e159 device=1
subvendor=8085
Jul 21 09:57:13 boldsoft kernel: wcfxs1: Wildcard TDM400P REV H port
0xe800-0xe8ff mem 0xfaffe000-0xfaffefff irq 18 at devic
e 9.0 on pci2
Jul 21 09:57:13 boldsoft kernel: FXS Attach for wcfxs1: deviceID :
0xe159
Jul 21 09:57:13 boldsoft kernel: Freshmaker version: 85
Jul 21 09:57:13 boldsoft kernel: 00 != 85
Jul 21 09:57:13 boldsoft kernel: 01 != 85
Jul 21 09:57:13 boldsoft kernel: 02 != 85
Jul 21 09:57:13 boldsoft kernel: 03 != 85
Jul 21 09:57:13 boldsoft kernel: 04 != 85
Jul 21 09:57:13 boldsoft kernel: 05 != 85
Jul 21 09:57:13 boldsoft kernel: 06 != 85
Jul 21 09:57:13 boldsoft kernel: 07 != 85
Jul 21 09:57:13 boldsoft kernel: 08 != 85
Jul 21 09:57:13 boldsoft kernel: 09 != 85

  ReplyForward


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-- Forwarded message --
From: Ganbold Tsagaankhuu [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com, asterisk-dev@lists.digium.com,
asterisk-bsd@lists.digium.com
Date: Tue, 2 Aug 2005 10:38:50 +0900
Subject: [Asterisk-Dev] Questions on Asterisk and CallerID
Hello,

I have few questions about Asterisk.
I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago.

1.I couldn't find Asterisk version using asterisk -V command.

How can I to find version information?

2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on
it.

I tried Asterisk CallerID feature, but unable to get it.
I tried callerid signaling V23, Bell202, DTMF, no success. Finally, I
found in our country (Mongolia) PSTN/Cellular provider send FSK/ETSI
type of CallerID.

Is Asterisk support such type of CallerID signaling?

If no, is there any way to get it?

3.I enjoyed Asterisk most of feature until now. I registered X-Pro
softphone, SIP analog and analog phone connected to FXS port too.

There one problem is I am unable to make outgoing call from SIP phone,
softphone, analog phone through FXO port.

Following is my Asterisk configuration:
--
zaptel.conf
loadzone=us
defaultzone=us
fxsks=1
fxoks=2


zapata.conf
context=bell
signaling=fxs_ks
group=1
channel = 1


context=home
group=2
signalling=fxo_ks
channel = 2


sip.conf
[]
type=friend
username=
;secret=
host=dynamic
nat=yes
defaultip=192.168.1.5
context=bell
reinvite=no
canreinvite=no
callerid=

[Asterisk-Users] Questions on Asterisk and CallerID

2005-08-01 Thread Ganbold Tsagaankhuu 
Hello,

I have few questions about Asterisk.
I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago.

1.I couldn't find Asterisk version using asterisk -V command.

How can I to find version information?

2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on
it.

I tried Asterisk CallerID feature, but unable to get it.
I tried callerid signaling V23, Bell202, DTMF, no success. Finally, I
found in our country (Mongolia) PSTN/Cellular provider send FSK/ETSI
type of CallerID.

Is Asterisk support such type of CallerID signaling?

If no, is there any way to get it?

3.I enjoyed Asterisk most of feature until now. I registered X-Pro
softphone, SIP analog and analog phone connected to FXS port too.

There one problem is I am unable to make outgoing call from SIP phone,
softphone, analog phone through FXO port.

Following is my Asterisk configuration:
--
zaptel.conf
loadzone=us
defaultzone=us
fxsks=1
fxoks=2


zapata.conf
context=bell
signaling=fxs_ks
group=1
channel = 1


context=home
group=2
signalling=fxo_ks
channel = 2


sip.conf
[]
type=friend
username=
;secret=
host=dynamic
nat=yes
defaultip=192.168.1.5
context=bell
reinvite=no
canreinvite=no
callerid=
[EMAIL PROTECTED]
allow=g729
allow=g723
allow=all


[]
type=friend
username=
;secret=
host=dynamic
nat=yes
defaultip=192.168.1.1
context=bell
reinvite=no
canreinvite=no
callerid=
[EMAIL PROTECTED]
allow=g729
allow=g723


extensions.conf
[bell]
exten = s,1,Wait
exten = s,2,Answer
exten = s,3,Playback(greetings)
exten = s,4,WaitExten


; used to record prompts
exten = 205,1,Wait(2)
exten = 205,2,Record(/tmp/greetings:alaw)
exten = 205,3,Wait(2)
exten = 205,4,Playback(/tmp/greetings)
exten = 205,5,Wait(2)
exten = 205,6,Hangup


exten = 111,1,Dial(CONSOLE/dsp)
exten = 111,2,Hangup


exten = 100,1,Answer
exten = 100,2,MusicOnHold()
exten = 100,4,Hangup


exten = 200,1,VoicemailMain


exten = 300,1,Dial(Zap/2)


exten = 400,1,Voicemail(9)


exten = 800,1,MeetMe(100|Mp)
exten = 800,2,Hangup


exten = 601,1,WaitMusicOnHold(30)


exten = 700,1,Dial(SIP/,20,rt)
exten = 900,1,Dial(SIP/,20,rt)


exten = _ZXXX,1,Answer
exten = _ZXXX,2,Dial(Zap/g1/${EXTEN})
exten = _Z,1,Answer
exten = _Z,2,Dial(Zap/g1/${EXTEN})
exten = _NX,1,Answer
exten = _NX,2,Dial(Zap/g1/${EXTEN})
exten = _NXXX,1,Answer
exten = _NXXX,2,Dial(Zap/g1/${EXTEN})


[home]
exten = s,1,Playback(greetings)
exten = 100,1,Answer
exten = 100,2,MusicOnHold()
exten = 100,4,Hangup


exten = 111,1,Dial(CONSOLE/dsp)
exten = 111,4,Hangup


exten = 700,1,Dial(SIP/,20,rt)
exten = 900,1,Dial(SIP/,20,rt)


exten = _ZXXX,1,Answer
exten = _ZXXX,2,Dial(Zap/g1/${EXTEN})
exten = _Z,1,Answer
exten = _Z,2,Dial(Zap/g1/${EXTEN})
exten = _NX,1,Answer
;exten = _NX,2,SetVar(TIMEOUT(AbsoluteTimeout)=10)
exten = _NX,3,Dial(Zap/g1/${EXTEN})
exten = _NXXX,1,Answer
exten = _NXXX,2,Dial(Zap/g1/${EXTEN})


I can to see following in /var/log/messages when I make outgoing call.


Jul 20 00:50:26 boldsoft kernel: Zapata Telephony Interface Registered
on major 196
Jul 20 00:50:26 boldsoft kernel: ZapTel device: vendor=e159 device=1
subvendor=8085
Jul 20 00:50:26 boldsoft kernel: wcfxo0: Wildcard X101P port
0xe800-0xe8ff mem 0xfaffe000-0xfaffefff irq 18 at device 9.0 on
pci2
Jul 20 00:50:26 boldsoft kernel: ZapTel Attach for wcfxo0: deviceID :
0xe159
Jul 20 00:50:26 boldsoft kernel: wcfxo: DAA mode is 'FCC'
Jul 20 00:50:26 boldsoft kernel: Found a Wildcard FXO: Wildcard X101P
Jul 20 00:50:26 boldsoft kernel: ZapTel device loaded.
Jul 20 00:50:33 boldsoft kernel: FXS device: vendor=e159 device=1
subvendor=b100
Jul 20 00:50:33 boldsoft kernel: wcfxs0: Wildcard TDM400P REV E/F
port 0xec00-0xecff mem 0xfafff000-0xfaff irq 17 at dev
ice 8.0 on pci2
Jul 20 00:50:33 boldsoft kernel: FXS Attach for wcfxs0: deviceID :
0xe159
Jul 20 00:50:33 boldsoft kernel: Freshmaker version: 63
Jul 20 00:50:33 boldsoft kernel: Freshmaker passed register test
Jul 20 00:50:35 boldsoft kernel: Module 0: Installed -- AUTO FXS
Jul 20 00:50:35 boldsoft kernel: ProSLIC sanity check failed
Jul 20 00:50:35 boldsoft kernel: Module 1: Not installed
Jul 20 00:50:35 boldsoft kernel: ProSLIC sanity check failed
Jul 20 00:50:35 boldsoft kernel: Module 2: Not installed
Jul 20 00:50:35 boldsoft kernel: ProSLIC sanity check failed
Jul 20 00:50:35 boldsoft kernel: Module 3: Not installed
Jul 20 00:50:35 boldsoft kernel: Found a Wildcard TDM: Wildcard
TDM400P
REV E/F (4 modules)
Jul 20 00:50:39 boldsoft kernel: Registered tone zone 0 (United States
/ North America)
Jul 21 02:36:28 boldsoft kernel: DIAL: T345598w
Jul 21 02:39:43 boldsoft kernel: DIAL: T345598w
Jul 21 02:45:35 boldsoft kernel: DIAL: T345598w
Jul 21 02:45:56 boldsoft kernel: DIAL: T99114909w
Jul 21 02:47:09 boldsoft kernel: DIAL: T345598w
Jul 21 02:47:56 boldsoft kernel: DIAL: T345595w
Jul 21 02:48:16 boldsoft

[Asterisk-Users] Re: oh323 compile problem in FreeBSD

2005-05-13 Thread Ganbold Tsagaankhuu 
: In member function `BOOL WrapMutex::Wait(const
char*, int, const char*, int)':
wrapper_misc.cxx:59: error: `PTimeInterval' undeclared (first use this function)
wrapper_misc.cxx:59: error: expected `;' before timeout
wrapper_misc.cxx:62: error: `timeout' undeclared (first use this function)
wrapper_misc.cxx:66: error: `cout' undeclared (first use this function)
wrapper_misc.cxx:66: error: 'class WrapMutex' has no member named 'Class'
wrapper_misc.cxx:66: error: `name' undeclared (first use this function)
wrapper_misc.cxx:66: error: `endl' undeclared (first use this function)
wrapper_misc.cxx:68: error: `PMutex' has not been declared
wrapper_misc.cxx:69: error: 'class WrapMutex' has no member named 'Class'
wrapper_misc.cxx:73: error: 'class WrapMutex' has no member named 'Class'
wrapper_misc.cxx: In member function `void WrapMutex::Signal(const
char*, int, const char*)':
wrapper_misc.cxx:81: error: `PMutex' has not been declared
wrapper_misc.cxx:81: error: no matching function for call to
`WrapMutex::Signal()'
wrapper_misc.cxx:80: note: candidates are: void
WrapMutex::Signal(const char*, int, const char*)
wrapper_misc.cxx:82: error: `cout' undeclared (first use this function)
wrapper_misc.cxx:82: error: 'class WrapMutex' has no member named 'Class'
wrapper_misc.cxx:82: error: `name' undeclared (first use this function)
wrapper_misc.cxx:82: error: `endl' undeclared (first use this function)
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/thread.h: At global scope:
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/thread.h:170: warning:
'void PX_ThreadEnd(void*)' declared `static' but never defined
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pprocess.h:147:
warning: 'void PXShowSystemWarning(PINDEX)' declared `static' but
never defined
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pprocess.h:148:
warning: 'void PXShowSystemWarning(PINDEX, int)' declared `static' but
never defined
gmake[1]: *** [wrapper_misc.o] Error 1
gmake[1]: Leaving directory `/usr/home/tsgan/asterisk-oh323-0.7.1/wrapper'
gmake: *** [subdirs_build] Error 1
asterisk#



On 5/9/05, Michael Manousos [EMAIL PROTECTED] wrote:
 
 Hi,
 
 I have never tried it on FreeBSD. Post it on the bugtracker and
 will see what can be done about it. Normally, just compilation
 flags and options need to change to make it work on FreeBSD.
 
 Regards,
 Michael.
 
 Ganbold Tsagaankhuu wrote:
  Hi,
 
  I'm trying to compile asterisk-oh323-0.7.1 in FreeBSD 5.3.
  I tried to use gmake but it exits with too many errors.
 
  Did you compile before oh323 in FreeBSD?
  How should I compile it under FreeBSD?
 
  thanks,
 
  Ganbold
 

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[Asterisk-Users] [Asterisk-Dev] Re: oh323 compile problem in FreeBSD

2005-05-13 Thread Ganbold Tsagaankhuu 
: In member function `BOOL WrapMutex::Wait(const
char*, int, const char*, int)':
wrapper_misc.cxx:59: error: `PTimeInterval' undeclared (first use this function)
wrapper_misc.cxx:59: error: expected `;' before timeout
wrapper_misc.cxx:62: error: `timeout' undeclared (first use this function)
wrapper_misc.cxx:66: error: `cout' undeclared (first use this function)
wrapper_misc.cxx:66: error: 'class WrapMutex' has no member named 'Class'
wrapper_misc.cxx:66: error: `name' undeclared (first use this function)
wrapper_misc.cxx:66: error: `endl' undeclared (first use this function)
wrapper_misc.cxx:68: error: `PMutex' has not been declared
wrapper_misc.cxx:69: error: 'class WrapMutex' has no member named 'Class'
wrapper_misc.cxx:73: error: 'class WrapMutex' has no member named 'Class'
wrapper_misc.cxx: In member function `void WrapMutex::Signal(const
char*, int, const char*)':
wrapper_misc.cxx:81: error: `PMutex' has not been declared
wrapper_misc.cxx:81: error: no matching function for call to
`WrapMutex::Signal()'
wrapper_misc.cxx:80: note: candidates are: void
WrapMutex::Signal(const char*, int, const char*)
wrapper_misc.cxx:82: error: `cout' undeclared (first use this function)
wrapper_misc.cxx:82: error: 'class WrapMutex' has no member named 'Class'
wrapper_misc.cxx:82: error: `name' undeclared (first use this function)
wrapper_misc.cxx:82: error: `endl' undeclared (first use this function)
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/thread.h: At global scope:
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/thread.h:170: warning:
'void PX_ThreadEnd(void*)' declared `static' but never defined
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pprocess.h:147:
warning: 'void PXShowSystemWarning(PINDEX)' declared `static' but
never defined
/usr/home/tsgan/pwlib/include/ptlib/unix/ptlib/pprocess.h:148:
warning: 'void PXShowSystemWarning(PINDEX, int)' declared `static' but
never defined
gmake[1]: *** [wrapper_misc.o] Error 1
gmake[1]: Leaving directory `/usr/home/tsgan/asterisk-oh323-0.7.1/wrapper'
gmake: *** [subdirs_build] Error 1
asterisk#



On 5/9/05, Michael Manousos [EMAIL PROTECTED] wrote:
 
 Hi,
 
 I have never tried it on FreeBSD. Post it on the bugtracker and
 will see what can be done about it. Normally, just compilation
 flags and options need to change to make it work on FreeBSD.
 
 Regards,
 Michael.
 
 Ganbold Tsagaankhuu wrote:
  Hi,
 
  I'm trying to compile asterisk-oh323-0.7.1 in FreeBSD 5.3.
  I tried to use gmake but it exits with too many errors.
 
  Did you compile before oh323 in FreeBSD?
  How should I compile it under FreeBSD?
 
  thanks,
 
  Ganbold
 

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[Asterisk-Users] oh323 compile problem in FreeBSD

2005-05-06 Thread Ganbold Tsagaankhuu 
Hi,

I'm trying to compile asterisk-oh323-0.7.1 in FreeBSD 5.3.
I tried to use gmake but it exits with too many errors.

Did somebody compile before oh323 in FreeBSD?
How should I compile it under FreeBSD?

thanks,

Ganbold
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[Asterisk-Users] Asterisk + GNUGK

2005-05-05 Thread Ganbold Tsagaankhuu 
Hi,

I'm trying to configure asterisk to work with gnugk-2.0.8. Something like:

SIP phones - ASTERISK - GNUGK -Cisco GW - PSTN
  |
   h323 phones


Following is h323.conf:

[general]
port = 1720
bindaddr = 0.0.0.0

disallow=all
allow=g729
gatekeeper = x.x.x.x
secret = 1234
AllowGKRouted = yes
noFastStart = yes
noH245Tunneling = yes
noSilenceSuppression = yes

[30598272]
type=h323
prefix=111,115,116,117
context=home
;e164=117

[115]
type=user
context=home
incominglimit=4

sip.conf

[general]
port=5060   ; Port to bind to
bindaddr=0.0.0.0; Address to bind SIP channel to
context=home; Default context for incoming calls
musicclass=default
;videosupport=yes
allow=g729
allow=g723

;externip = 202.179.0.164
;localnet=192.168.0.0/255.255.0.0


[111]
type=friend
username=111
;secret=
host=dynamic
nat=yes
defaultip=192.168.0.11
context=home
canreinvite=no
callerid=111
[EMAIL PROTECTED]

[112]
type=friend
username=112
;secret=
nat=yes
host=dynamic
context=home
canreinvite=no
callerid=112
[EMAIL PROTECTED]

[115]
type=friend
username=115
;secret=1234
defaultip=192.168.0.62
nat=yes
host=dynamic
context=home
canreinvite=no
callerid=115
[EMAIL PROTECTED]

[116]
type=friend
username=116
;secret=4321
host=dynamic
context=home
canreinvite=no
callerid=116
[EMAIL PROTECTED]


As in above configuration I'm registering Asterisk as an endpoint to gnugk.
It is working and I can make calls from SIP phones to PSTN.
However my question is, how can I call from h323 endpoints to SIP
phones or vice versa in above case?
Is it possible? I'm afraid, it can't since asterisk is itself an one
endpoint to gnugk.
If possible how can I make it work?

If not, is it possible to register or make each SIP phones to be known to gnugk?
How can I accomplish that? Ideally this solution could be the best.

It would be very helpful if somebody can show me the config samples.

thanks in advance,

Ganbold
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Re: [Asterisk-Users] Asterisk + GNUGK

2005-05-05 Thread Ganbold Tsagaankhuu 
Hi,

On 5/5/05, Niksa Baldun [EMAIL PROTECTED] wrote:
 Assuming your h.323 phones are registered with gnugk, you need to
 instruct gnugk to forward certain numbers to Asterisk. In OH323 (which I
 am using) you would need to add something like:
 
 [register]
 gwprefix=0
 gwprefix=1
 etc.
 
 In h323.conf, I believe you have to add prefix=xxx in your endpoint
 definition.

I tried to put prefixes in h323.conf in following way, however it didn't work:

[30598272]
type=h323
prefix=111
prefix=115
prefix=116
prefix=117
context=home
;e164=117
--

Am I doing something wrong? Does somebody have configuration samples?

thanks,

Ganbold


 
 Bear in mind though that H.323 support in Asterisk is rather inadequate
 (only basic telephony functions are available).
 
 Niksa Baldun
 
 
 Ganbold Tsagaankhuu wrote:
 
 Hi,
 
 I'm trying to configure asterisk to work with gnugk-2.0.8. Something like:
 
 SIP phones - ASTERISK - GNUGK -Cisco GW - PSTN
   |
h323 phones
 
 
 Following is h323.conf:
 
 [general]
 port = 1720
 bindaddr = 0.0.0.0
 
 disallow=all
 allow=g729
 gatekeeper = x.x.x.x
 secret = 1234
 AllowGKRouted = yes
 noFastStart = yes
 noH245Tunneling = yes
 noSilenceSuppression = yes
 
 [30598272]
 type=h323
 prefix=111,115,116,117
 context=home
 ;e164=117
 
 [115]
 type=user
 context=home
 incominglimit=4
 
 sip.conf
 
 [general]
 port=5060   ; Port to bind to
 bindaddr=0.0.0.0; Address to bind SIP channel to
 context=home; Default context for incoming calls
 musicclass=default
 ;videosupport=yes
 allow=g729
 allow=g723
 
 ;externip = 202.179.0.164
 ;localnet=192.168.0.0/255.255.0.0
 
 
 [111]
 type=friend
 username=111
 ;secret=
 host=dynamic
 nat=yes
 defaultip=192.168.0.11
 context=home
 canreinvite=no
 callerid=111
 [EMAIL PROTECTED]
 
 [112]
 type=friend
 username=112
 ;secret=
 nat=yes
 host=dynamic
 context=home
 canreinvite=no
 callerid=112
 [EMAIL PROTECTED]
 
 [115]
 type=friend
 username=115
 ;secret=1234
 defaultip=192.168.0.62
 nat=yes
 host=dynamic
 context=home
 canreinvite=no
 callerid=115
 [EMAIL PROTECTED]
 
 [116]
 type=friend
 username=116
 ;secret=4321
 host=dynamic
 context=home
 canreinvite=no
 callerid=116
 [EMAIL PROTECTED]
 
 
 As in above configuration I'm registering Asterisk as an endpoint to gnugk.
 It is working and I can make calls from SIP phones to PSTN.
 However my question is, how can I call from h323 endpoints to SIP
 phones or vice versa in above case?
 Is it possible? I'm afraid, it can't since asterisk is itself an one
 endpoint to gnugk.
 If possible how can I make it work?
 
 If not, is it possible to register or make each SIP phones to be known to 
 gnugk?
 How can I accomplish that? Ideally this solution could be the best.
 
 It would be very helpful if somebody can show me the config samples.
 
 thanks in advance,
 
 Ganbold
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[Asterisk-Users] Mysql/Radius Authentication

2005-05-04 Thread Ganbold Tsagaankhuu 
Hi all,

I'm using asterisk-1.0.7. I need to configure asterisk in such way
that it authenticates users from mysql DB. Is it possible to
authenticate SIP users from mysql database?
It seems to me that chan_sip2 code from Olle E. Johansson, Edvina.net,
[EMAIL PROTECTED] can authenticate users from mysql. However I looked for
it everywhere and didn't find. Where can I download chan_sip2 code?
Is there any other way I can authenticate SIP users from mysql in asterisk?
Is it possible to make asterisk work with radius?
I appreciate if somebody can give me some hints and advices in this regard.

thanks in advance,

Ganbold
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