Re: [asterisk-users] IAX configuration

2007-09-28 Thread yonoko molomo 
Hi,
i think that is not the point.
the call works, what is not working is the IAX config.
somehow i need to put manually all users of the foreign asterisk
(user, password...).

if i put type=friend, it does not work in any case.
if i put type=peer it works only if i define the users of the foreign
asterisk and also an entry for the foreign server

that should not be the normal behaviour, i guess.

any ideas?
thanks



2007/9/27, Mojo with Horan  Company, LLC [EMAIL PROTECTED]:
 when using variables, use ${variablename} instead of $(variablename) --
 (squiggly braces instead of parentheses) -- I'm not sure parentheses are
 allowed.

 yonoko molomo wrote:
  Now I update the extensions.conf file accordingly.
  exten = clientA_Number,1,Dial(sip/$(exten),10)

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[asterisk-users] IAX configuration

2007-09-27 Thread yonoko molomo 
Hi,

I have some problems and doubts connecting two asterisk servers.

I have one asterisk (serverA), with 1 sip client registered (clientA).
I have another asterisk (sever B), with another client (clientB).
Now I want to call from client A to B and from B to A.

Searching in google i find many configuration examples. For instance:
http://etel.wiki.oreilly.com/wiki/index.php/Peering_two_Asterisk_servers_using_IAX
There are hundreds of pages like this. Also the book Asterisk the
future of telephony describes how to configure IAX.
But it seems that there are many things missing.

I understand I need to configure iax.conf in server A and add an entry
for server B as peer.
I also need to include a register line in iax.conf of server B. If i
do this, i have server B registered in server A.

I do the same in server B.

The problem is that I see some rejected messages, and the servers
are not registering. I do not see anything else that i need to do in
the configuration examples.
Finally, after doing a lot of tests, I tried adding in users.conf
some lines like:
[serverA]
type=peer
callwaiting = yes
fullname = serverA
hasagent = yes
hasdirectory = no
hasiax = yes
hasmanager = no
hassip = no
hasvoicemail = yes
host = dynamic
secret = serverA
threewaycalling = yes
vmsecret = 1234
registeriax = yes
registersip = no
canreinvite = no
nat = no
dtmfmode = rfc2833

im not sure of some of these parameter. but basicalli i define a new
user for the server. if i do not define a user like this in users.conf
the registration of the server is rejected.

I type in the CLI iax2 show registry
and i see that server A is registered in B and B in A. so it seems fine.

Now I update the extensions.conf file accordingly.
in server A i define something like
exten = clientA_Number,1,Dial(sip/$(exten),10)
exten = clientA_Number,2,hangup()
exten = clientB_Number,1,Dial(iax2/serverB,$(exten))
exten = clientB_Number,2,hangup()

i do the similar in serverB.

i am very frustrated because the call does not work. the error is Call
rejected by x.x.x.x  No authority found

Again, after doing a lot of tests, i noticed that if i add a new user
in iax.conf like:
[clientB]
type=friend
username=clientB
secret=userB
...

and i do the following in serverB's iax.conf:
register = clientB:[EMAIL PROTECTED]

then it seems to work.
but i have some doubts on it.
i have:
- clientA registered in serverA
- clientB registered in serverB
- serverA registered in serverB
- serverB registered in serverA

then, why do i need to do an explicit register of clientB in server A
in the iax.conf and in the users.conf?
i am not sure if this is mandatory (i did not find this configuration
in any example, but if i do not do this, the call is rejected in the
remote asterisk server)

why should server A know all users in server B?
is it not enough if server B is registered in server A?

i am not an expert in asterisk and i am not sure if this is the
correct configuration, but i am a bit tired testing and this is the
only way i found to make the calls work, but i dont find it is
reasonable, especially when the system grows and then my asterisk
server peers with many other servers (do i need to configure my
users.conf and iax.conf for *each* of the users of other asterisks?)

could someone give me some hints?
thanks

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Re: [asterisk-users] IAX configuration

2007-09-27 Thread yonoko molomo 
hi,
it does not help.
at first i already tried using type=friend.
but i am not able to make calls.
in the 'caller' asterisk get:

WARNING[18541]: chan_iax2.c:7101 socket_process: Call rejected by
x.x.x.x: No authority found
-- Hungup 'IAX2'

in the 'called' asterisk i get following error:
chan_iax2.c:7584 socket_process: Host x.x.x.x failed to authenticate as serverB

but in the CLI, i type iax2 show registry and i see that servers are registered.
Host  dnsmgr  UsernamePerceived Refresh  State
x.x.x.x:4569   N   servery.y.y.y:4569 60  Registered
i asume registered means also authenticated but it complaints.


I already followed that link but it does not work. i need to change some things.
first, i need to change friend to peer. if i use type=friend it does
not work for me.

then i need to define all users of the foreign asterisk, which is not
normal, i guess.

I also need to define the ip address of the servers as host=dynamic.
if i put host=192.168.1.10 or similar, i get the message in the CLI:
Peer 'server' is not dynamic (from 192.168.1.10)

somehow i feel that there is something missing in the configuracion
examples, but i do not know what is it.
any ideas?

2007/9/27, Gordon Henderson [EMAIL PROTECTED]:
 On Thu, 27 Sep 2007, Anthony Messina wrote:

  On Thursday 27 September 2007 09:23:09 am yonoko molomo wrote:
  Hi,
 
  I have some problems and doubts connecting two asterisk servers.
 
  I have one asterisk (serverA), with 1 sip client registered (clientA).
  I have another asterisk (sever B), with another client (clientB).
  Now I want to call from client A to B and from B to A.
 
  Searching in google i find many configuration examples. For instance:
  http://etel.wiki.oreilly.com/wiki/index.php/Peering_two_Asterisk_servers_us
  ing_IAX There are hundreds of pages like this. Also the book Asterisk the
  future of telephony describes how to configure IAX.
  But it seems that there are many things missing.
 
  I understand I need to configure iax.conf in server A and add an entry
  for server B as peer.
  I also need to include a register line in iax.conf of server B. If i
  do this, i have server B registered in server A.
 
  I do the same in server B.
 
 
  you'll need to define each as a friend of each other.

 Start here:

http://astrecipes.net/index.php?n=204

 Gordon

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Re: [asterisk-users] Cisco 7940G licensing with asterisk

2007-09-27 Thread yonoko molomo 
Hi,
i bought this device and the cost of the 7040G itself was similar to
the license. if im not wrong, the telephone cost around 80€. the sip
license was around 80€ as well

however, i am quite annoyed because the phone did not come with sip,
but callmanager so i cant use it as i planned.
i have read somewhere that I need to change the firmware, but i
require a cisco account to download the firmware (but nobody provided
me this account). we paid for the SIP license, but we did not get a
SIP-capable device, and we do not have the way to download the
firmware (yet).

Regarding the power adapter, I had to buy them sepparately. since i do
not have POE devices i cant answer your last question.



2007/9/27, Erick Perez [EMAIL PROTECTED]:
 Hi there,
 In Cisco web site
 http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
 It says that regardless of the technology used you have to buy a licencse.
 Does the license apply to use the phone with asterisk, or, can i just
 buy the phone?

 Also, the phone does not requiere to use an AC adapter if used with
 PoE injectors/switches.
 Can non-Cisco PoE injectors/switches be used with this phone?

 Thanks,

 --
 
 Erick Perez
 

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Re: [asterisk-users] res_snmp

2007-09-19 Thread yonoko molomo 
Hi,
Thanks for the answer.

 Yes, res_snmp seems to be sensitive to the specific version of net_snmp.
 I wrote some notes on this - see 
 http://www.voip-info.org/wiki/view/Asterisk+monitoring

 Basically I ended up installing netsnmp from source, and things started 
 working.
what do you mean? you compile net-snmp from source? which version do
you compile?

 I'm currently writing a little demo program which lets you see calls via snmp
 but I'm a bit stuck on the graphical representation.

 Which snmp tools will you use to monitor asterisk ?
I still dont know which tool use to monitor asterisk, probably nagios.
thanks


 Tim.

 - Original Message -
 From: yonoko molomo [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Sent: Wednesday, September 12, 2007 4:51:11 PM (GMT) Europe/London
 Subject: [asterisk-users] res_snmp

 Hi,

 I have problems compiling asterisk 1.4.11 with res_snmp.
 I do 'make menuselect', and I see that this resource module depends on 
 netsnmp.
 I am using centOS 4.5.
 I do:
  yum install net-snmp net-snmp-devel net-snmp-utils net-snmp-libs
 I don't know if i am missing something.

 I go to the source directory and I do:
 ./configure

 but still does not work:
  ...
  checking for curses.h... (cached) yes
  checking for net-snmp-config... /usr/bin/net-snmp-config
  checking for snmp_register_callback in -lnetsnmp... no
  ...

 When i run 'make menuselect' I cannot get rid of the XXX, and i can't
 select res_snmp.

 I also tried to install net-snmp-perl package but it does not help. i
 have no clue how to continue.

 any ideas?
 thanks

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[asterisk-users] res_snmp

2007-09-12 Thread yonoko molomo 
Hi,

I have problems compiling asterisk 1.4.11 with res_snmp.
I do 'make menuselect', and I see that this resource module depends on netsnmp.
I am using centOS 4.5.
I do:
 yum install net-snmp net-snmp-devel net-snmp-utils net-snmp-libs
I don't know if i am missing something.

I go to the source directory and I do:
./configure

but still does not work:
 ...
 checking for curses.h... (cached) yes
 checking for net-snmp-config... /usr/bin/net-snmp-config
 checking for snmp_register_callback in -lnetsnmp... no
 ...

When i run 'make menuselect' I cannot get rid of the XXX, and i can't
select res_snmp.

I also tried to install net-snmp-perl package but it does not help. i
have no clue how to continue.

any ideas?
thanks

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Re: [asterisk-users] H.323

2007-08-03 Thread yonoko molomo 
Hi,
I have used h323, oh323 and ooh323.
My experience is that ooh323 does not work properly, i dont recommend it.
I dont know why, but the sound is bad, with sound breaks. I also need
to put some wait (2) functions after the answer( ) or playback( )
functions, it think that asterisk takes some time to stablish the
ooh323 channel (maybe it is due to other reason, i dont know exactly)
but during this time no sound is played, so the first seconds of
conversation or playback are cutted. ooh323 did not work for me at
all.

oh332 worked fine in asterisk 1.2 (i did not try in 1.4, i guess it is fine).

h323 works fine in asterisk 1.4. it is the one i am using now, and i
have no problems with it.

bye now

2007/8/2, Rurouni Alucard [EMAIL PROTECTED]:
 Hi there,

 I have use the H.323 module that comes with asterisk-addons and i
 consider it (so far) VERY stable for my needs.
 Im talking about 10,000 minutes at month , + or - , and never had a
 crash or something bad about it.

 Personally, i recommend it,


 --
 J. P.
 rakh at slackware-es dot com

 bilal ghayyad wrote:
  Hi List;
 
  Did any one tried the H.323 module? How much it is
  stable and work fine?
 
  Regards,
  
  ITS
  IP Telephony and Contact Center Engineer
  Eng. Bilal Ghayad
  Mobile: 00965 9849460
 
 
 
  Ready
   for the edge of your seat?
  Check out tonight's top picks on Yahoo! TV.
  http://tv.yahoo.com/
 
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[asterisk-users] radius support

2007-08-02 Thread yonoko molomo 
hi,
how to add radius support to asterisk 1.4.5?
i do make menuselect and i do not see any module or option related to
radius, pam, authenticacion or similar.
any ideas?
thanks

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Re: [asterisk-users] radius support

2007-08-02 Thread yonoko molomo 
Hi,

 http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html



Thanks, I have already seen that document before but it did not help
much to have a better understanding to set up radius with asterisk.

In 4.3 it is written: Asterisk has been patched along with the
previously decribed PAM radius module. But I was not able to find
that patch. In any case, is pam_radius not supported by asterisk
without patching it? I thought it was supported (i am using stable
version 1.4.5 of asterisk)

In the next sentence, A discussion on how to provide RADIUS functions
to Asterisk can be found here, along with the patch :
http://bugs.digium.com/view.php?id=5424; there is that link, but
again, it is not helping me to understand how to do this.

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[asterisk-users] users in sip.conf or in users.conf? what is the difference?

2007-07-23 Thread yonoko molomo 
Hi,

I have a question which might be a bit simple for advanced users.
Where is the correct config file to define users in asterisk 1.4?

I was using sip.conf file and it was working fine.
Recently I installed an asterisk web interface and I see that new
users created using this web interface are added into the users.conf
file, and not into sip.conf

What is the difference between using sip.con or users.conf file to
configure sip users?

So I decided that it makes sense to put user definition in the
users.conf file. But sometimes I noticed that using users.conf
asterisk complains with some messages like:
Registration from 'sip:[EMAIL PROTECTED]' failed for 'xxx.xxx.xxx.xxx'
- Device does not match ACL

Now I'm not sure if there was something missing in the definition of
the user in users.conf file, but copying *exactly* the same user
definition lines in sip.conf was solving the problem.

Any idea?
Thanks

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Re: [asterisk-users] Music on hold problem

2007-07-19 Thread yonoko molomo 
Hi,

Thanks for the answer.
Actually I noticed that several things dont work properly and I think
the ooh323 channel driver is the reason.

For instance, when I configure my extensions.conf to answer the phone
and playback a sound (for instance nobodyavailable message after
10secs), only the last part of the sound is played (i hear sometimes
only the last word, sometimes i hear nothing!). In the CLI I see that
asterisk is playing the sound file, but in the h323 phone i do not
hear the complete message.
I solved this issue putting a wait(3) just before the Playback
function. It seems that somehow ohh323 or asterisk needs some time to
setup the channel or something, and the sound has been already played
in the meantime.

I also have the same sound problem using the Meetme function to join a
conference bridge. I should hear After the tone say your name and
then press the pound key. I normally do not hear the first 4 or 5
words. I tried to put some wait functions, but here it does not work.
In the CLI I see again those strange messages (Don't know how to
indicate condition -xxx on ooh323c_1...), and asterisk says that he is
playing several sound files, but in the phones

The same problem using Voicemail. I should hear a message asking for
my mailbox number, but normally I do not hear anything.

Do you think the patch I will fix the problems?

I will try later, thanks

2007/7/19, Russell Bryant [EMAIL PROTECTED]:
 yonoko molomo wrote:
  [Jul 17 11:19:22] WARNING[23645]: src/chan_h323.c:1044
  ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_1
  [Jul 17 11:19:25] NOTICE[23645]: rtp.c:783 process_rfc3389: Comfort
  noise support incomplete in Asterisk (RFC 3389). Please turn off on
  client if possible. Client IP: 10.4.0.116
  [Jul 17 11:19:25] WARNING[23645]: src/chan_h323.c:1044
  ooh323_indicate: Don't know how to indicate condition 16 on ooh323c_1

 This would be a bug in the ooh323 channel driver.  Feel free to report it to
 bugs.digium.com.  I think it's an easy fix.

 ...

 In fact, here is a patch that should fix it.  Feel free to go ahead and give 
 it
 a try and let me know if it fixes the problem for you.  However, please still
 report it to bugs.digium.com, or I will forget to merge the change.


 Index: asterisk-ooh323c/src/chan_h323.c
 ===
 --- asterisk-ooh323c/src/chan_h323.c(revision 413)
 +++ asterisk-ooh323c/src/chan_h323.c(working copy)
 @@ -1036,9 +1036,16 @@
   ast_set_flag(p, H323_ALREADYGONE);
}
break;
 +  case AST_CONTROL_HOLD:
 + ast_moh_start(ast, data, NULL);
 + break;
 +  case AST_CONTROL_UNHOLD:
 +ast_moh_stop(ast);
 + break;
 case AST_CONTROL_PROCEEDING:
 case AST_CONTROL_RINGING:
 case AST_CONTROL_PROGRESS:
 +  case -1;
break;
 default:
ast_log(LOG_WARNING,Don't know how to indicate condition %d on 
 %s\n,


 --
 Russell Bryant
 Software Engineer
 Digium, Inc.

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Re: [asterisk-users] asterisk 1.4 and gnugk with ooh323

2007-07-17 Thread yonoko molomo 
Hi,
Thanks for the answer.
Yes, I think both channels are built.
I see following messages at startup:
 Parsing '/etc/asterisk/h323.conf': Found
Creating H.323 Endpoint
 Parsing '/etc/asterisk/users.conf': Found
 Registered channel type 'H323' (The NuFone Network's Open H.323 Channel Driver)
 H.323 listener started chan_h323.so =
 The NuFone Network's OpenH323 Channel Driver

[and few lines after]
 Parsing '/etc/asterisk/ooh323.conf': Found
 Registered channel type 'OOH323' (Objective Systems H323 Channel Driver)
chan_ooh323.so = (^[[33;40mObjective Systems H323 Channel^[[0;37;40m)

I am trying to use only ooh323 (removing the h323.conf file) but still
does not work.
I believe it is a configuration problem.
Can someone point me to the correct configuration or example? The ones
I have found so far did not help me much.

Thanks

2007/7/16, Dovid B [EMAIL PROTECTED]:
 There are different h323 channel drivers. You seem to have built both. If
 you try to dial using Dial(h323/.) then you need h323.conf. If you try
 dialing using Dial(ooh323/.) then you need ooh323.conf. For me
 personally the ooh323 (can't remember the name of it  - sleep deprivation)
 works better for me.

 - Original Message -
 From: yonoko molomo [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Sent: Monday, July 16, 2007 6:13 PM
 Subject: [asterisk-users] asterisk 1.4 and gnugk with ooh323


  Hello all,
  I have seen some people asking how to configure asterisk to work with
  h323 but i did not manage to do fix it yet (i am not an asterisk
  expert).
  Can someone help me configuring asterisk?
  It is already compiled asterisk 1.4.5 with H323 support.
  Everything looks fine.
  Then i understand i need to configure several files:
  -sip.conf
  -ooh323.conf
  -extensions.conf
 
  do i also need to configure the h323.conf?
 
  i want asterisk and gnugk in the same machine (lets say ip 192.168.0.10).
  then sip_client at 192.168.0.100 (telephone number assigned for
  instance 100) and H323_client at 192.168.0.200 (telephone number
  assigned 200)
 
  how do i configure the files to do this?
 
  if i type show channeltypes i see:
 
  TypeDescription  Devicestate
  Indications  Transfer
  --  ---  ---
  ---  
  OOH323  Objective Systems H323 Channel Driverno   yes
 no
  SIP Session Initiation Protocol (SIP)yes  yes
 yes
  H323The NuFone Network's Open H.323 Channel  no   yes
 no
 
  What does devicestate and trasfer mean? it is set to no. is this ok?
 
 
  if someone could show me how to  configure the files to do this i
  would be very grateful.
 
  thanks!
 
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Re: [asterisk-users] asterisk 1.4 and gnugk with ooh323

2007-07-17 Thread yonoko molomo 
hi,
i fixed the problem.
as i thought it was a configuration problem, i was not defining the
asterisk users at ooh323.conf.
now it seems to work,
thanks

2007/7/17, Dovid B [EMAIL PROTECTED]:
 What output do you get from the CLI ?

 - Original Message -
 From: yonoko molomo [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Tuesday, July 17, 2007 9:59 AM
 Subject: Re: [asterisk-users] asterisk 1.4 and gnugk with ooh323


  Hi,
  Thanks for the answer.
  Yes, I think both channels are built.
  I see following messages at startup:
  Parsing '/etc/asterisk/h323.conf': Found
  Creating H.323 Endpoint
  Parsing '/etc/asterisk/users.conf': Found
  Registered channel type 'H323' (The NuFone Network's Open H.323 Channel
  Driver)
  H.323 listener started chan_h323.so =
  The NuFone Network's OpenH323 Channel Driver
 
  [and few lines after]
  Parsing '/etc/asterisk/ooh323.conf': Found
  Registered channel type 'OOH323' (Objective Systems H323 Channel Driver)
  chan_ooh323.so = (^[[33;40mObjective Systems H323 Channel^[[0;37;40m)
 
  I am trying to use only ooh323 (removing the h323.conf file) but still
  does not work.
  I believe it is a configuration problem.
  Can someone point me to the correct configuration or example? The ones
  I have found so far did not help me much.
 
  Thanks
 
  2007/7/16, Dovid B [EMAIL PROTECTED]:
  There are different h323 channel drivers. You seem to have built both. If
  you try to dial using Dial(h323/.) then you need h323.conf. If you
  try
  dialing using Dial(ooh323/.) then you need ooh323.conf. For me
  personally the ooh323 (can't remember the name of it  - sleep
  deprivation)
  works better for me.
 
  - Original Message -
  From: yonoko molomo [EMAIL PROTECTED]
  To: asterisk-users@lists.digium.com
  Sent: Monday, July 16, 2007 6:13 PM
  Subject: [asterisk-users] asterisk 1.4 and gnugk with ooh323
 
 
   Hello all,
   I have seen some people asking how to configure asterisk to work with
   h323 but i did not manage to do fix it yet (i am not an asterisk
   expert).
   Can someone help me configuring asterisk?
   It is already compiled asterisk 1.4.5 with H323 support.
   Everything looks fine.
   Then i understand i need to configure several files:
   -sip.conf
   -ooh323.conf
   -extensions.conf
  
   do i also need to configure the h323.conf?
  
   i want asterisk and gnugk in the same machine (lets say ip
   192.168.0.10).
   then sip_client at 192.168.0.100 (telephone number assigned for
   instance 100) and H323_client at 192.168.0.200 (telephone number
   assigned 200)
  
   how do i configure the files to do this?
  
   if i type show channeltypes i see:
  
   TypeDescription  Devicestate
   Indications  Transfer
   --  ---  ---
   ---  
   OOH323  Objective Systems H323 Channel Driverno   yes
  no
   SIP Session Initiation Protocol (SIP)yes  yes
  yes
   H323The NuFone Network's Open H.323 Channel  no   yes
  no
  
   What does devicestate and trasfer mean? it is set to no. is this ok?
  
  
   if someone could show me how to  configure the files to do this i
   would be very grateful.
  
   thanks!
  
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[asterisk-users] Music on hold problem

2007-07-17 Thread yonoko molomo 
Hi,
I am using asterisk 1.4.
I have confgured the musiconhold.conf file.
However, when i make a call and then hold the call it does nothing.
in the CLI i do not see the starting/stopping musiconhold messages.

i am making calls from sip to h323 using asterisk assip/h323 gateway
(with gnugk and ooh323).
i get the following messages when putting the call on hold:

-- Executing [EMAIL PROTECTED]:1] Ringing(OOH323/1169fed2-70c5, ) in new 
stack
-- Executing [EMAIL PROTECTED]:2] Dial(OOH323/1169fed2-70c5,
SIP/ht04|6|r) in new stack
-- Called ht04
-- SIP/ht04-081fe7f0 is ringing
-- SIP/ht04-081fe7f0 answered OOH323/1169fed2-70c5
...on hold...
[Jul 17 11:19:22] WARNING[23645]: src/chan_h323.c:1044
ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_1
[Jul 17 11:19:25] NOTICE[23645]: rtp.c:783 process_rfc3389: Comfort
noise support incomplete in Asterisk (RFC 3389). Please turn off on
client if possible. Client IP: 10.4.0.116
[Jul 17 11:19:25] WARNING[23645]: src/chan_h323.c:1044
ooh323_indicate: Don't know how to indicate condition 16 on ooh323c_1

I have no idea why the musiconhold is not triggered,  what those
messages mean (dont know how to indicate condition) and if they are
related to the music on hold problem


someone has any idea?

in the CLI i type
CLI moh show files
and i see one file i put in the directory.

i tried configuring the musiconhold.conf file as quitemp3, files and
also custom (installing mpg123) but none of them starts musiconhold.

do i need to activate musiconhold somewhere else?

any help is welcome
thanks

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[asterisk-users] asterisk 1.4 and gnugk with ooh323

2007-07-16 Thread yonoko molomo 
Hello all,
I have seen some people asking how to configure asterisk to work with
h323 but i did not manage to do fix it yet (i am not an asterisk
expert).
Can someone help me configuring asterisk?
It is already compiled asterisk 1.4.5 with H323 support.
Everything looks fine.
Then i understand i need to configure several files:
-sip.conf
-ooh323.conf
-extensions.conf

do i also need to configure the h323.conf?

i want asterisk and gnugk in the same machine (lets say ip 192.168.0.10).
then sip_client at 192.168.0.100 (telephone number assigned for
instance 100) and H323_client at 192.168.0.200 (telephone number
assigned 200)

how do i configure the files to do this?

if i type show channeltypes i see:

TypeDescription  Devicestate
Indications  Transfer
--  ---  ---
---  
OOH323  Objective Systems H323 Channel Driverno   yes
no
SIP Session Initiation Protocol (SIP)yes  yes
yes
H323The NuFone Network's Open H.323 Channel  no   yes
no

What does devicestate and trasfer mean? it is set to no. is this ok?


if someone could show me how to  configure the files to do this i
would be very grateful.

thanks!

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