[asterisk-users] FATAL: Module wcdtm not found.

2007-08-30 Thread jonny hashem
Hi:
I purchased TDM40B from a week ago from digium, iam
trying to install it to the current version of zaptel
(zaptel-1.4.5), but when i make modprobe wctdm these
came out :
[EMAIL PROTECTED] root]# modprobe wctdm
FATAL: Module wctdm not found.
FATAL: Error running install command for wctdm




  

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[asterisk-users] FATAL: Module wcdtm not found

2007-08-30 Thread jonny hashem
hi:
Iam using Mandrake 10.1 ,with kernel 2.6.8.1-12mdk 
and here where are zaptel.ko:
/lib/modules/2.6.8.1-12mdk/misc/zaptel.ko
/lib/modules/2.6.8.1-12mdkcustom/extra/zaptel.ko

thanks
jonny


   

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[asterisk-users] softphone with g729 codec

2007-07-02 Thread jonny hashem
Hi: 
  Iam looking for a sip softphone that supports g729 codec
Any one have an idea ?
 
   
   Reagrds;
  jonnyhashem  


 
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[Asterisk-Users] Steps to make trunked iax2

2006-04-07 Thread jonny hashem
Hi:
Is the difference of Asterisk verisons on two servers
effect on the iax2 trunking between them ?

Thanks;
jonny 

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[Asterisk-Users] Steps to make trunked iax2

2006-04-06 Thread jonny hashem
Hi:
What are the steps to make a trunked iax2 ?

Thanks in advance; 

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[Asterisk-Users] TDM04B sound volume

2006-03-30 Thread jonny hashem
HI:
Is there any way to raise up sound volume on fxo on
TDM04B without changing tx-gain and rx-gain ?

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[Asterisk-Users] show calls

2006-02-16 Thread jonny hashem
HI:
what is command on console to know how many calls are
sending in the same time?

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[Asterisk-Users] Aterisk 1.2.1 zaptel module not found

2005-12-30 Thread jonny hashem
Hi:
i have compiled Asterisk 1.2.1 without any problems
,But when i've tried to load the zaptel modules by
making modprobe zaptel this message shown:
FATAL: Module zaptel not found.

Regards;
jonny


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[Asterisk-Users] Best Voip provider

2005-12-24 Thread jonny hashem
Hi list:
i have a bad experience with voip providers , Any body
knows a voip provider i can depend on and to trust
with good rates and quality?




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[Asterisk-Users] astcc issue

2005-12-15 Thread jonny hashem
Hi list:
I need to create a routes list to specific card number
wih different prices than the initial routes list
,because markup donot achieve my purpose and markup
use for changing prices for all routes,and i need to
change prices for specific routes. So is there any
possible way to do that?

Regards;
jonny


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[Asterisk-Users] Iax2 connection failed

2005-12-03 Thread jonny hashem
HI:
i tried to send calls to callshopcompany
(www.callshopcompany.com) using iax2 but the call
fails giving me this:
 
dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
iax2/callshopcompany/0017046872001) in new stack
-- Called callshopcompany/0017046872001
-- Call accepted by 213.61.187.150 (format g729)
-- Format for call is g729
-- Hungup 'IAX2/callshopcompany/2'
  == No one is available to answer at this time
Dec  3 11:09:21 WARNING[23053]: pbx.c:1949
ast_pbx_run: Timeout, but no rule 't' in context
'calls'
  Hangup on console 

i have this my extensions.conf:
[calls]
exten = _00.,1,Dial,iax2/callshopcompany/${EXTEN}
;
i have this my iax.conf :
[callshopcompany]
type=peer
host=213.61.187.150
username=X
secret=X
disallow=all
allow=g729





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[Asterisk-Users] cannot dial on console on asterisk 1.2

2005-12-01 Thread jonny hashem
HI:
I ve downloded asterisk 1.2 and when i tried to dial
on console this message appears:
No such command 'dial' (type 'help' for help)

Despite iam not running any audio softwares.

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[Asterisk-Users] callback.agi script

2005-11-26 Thread jonny hashem
Hi list:
Where i have to go to get  callback.agi script i
searched every where but i didnt find it. 




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[Asterisk-Users] problems with callback.agi script

2005-11-26 Thread jonny hashem
Hi:
i used this callback.agi script and  here what i ve
get :
  Executing Wait(OSS/dsp, 1) in new stack
-- Executing AGI(OSS/dsp, callback.agi) in new
stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/callback.agi
Use of uninitialized value in string ne at
/var/lib/asterisk/agi-bin/callback.agi line 56,
STDIN line 14.
Nov 26 16:58:43 WARNING[8278]: file.c:475
ast_openstream: File
/var/lib/asterisk/agi-bin/get-phonenumber does not
exist in any format
Nov 26 16:58:43 WARNING[8278]: file.c:787
ast_streamfile: Unable to open
/var/lib/asterisk/agi-bin/get-phonenumber (format
slin): No such file or directory
MYCALLBACK: User Hungup (-1)
-- AGI Script callback.agi completed, returning 0
-- Executing Hangup(OSS/dsp, ) in new stack


what does this line means and what i have to change: 
Use of uninitialized value in string ne at
/var/lib/asterisk/agi-bin/callback.agi line 56,
STDIN line 14.


and in which directory i should put the sound files?




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[Asterisk-Users] problems with callback.agi script

2005-11-26 Thread jonny hashem
Hi :
i have the required soundfiles and i put it in
/var/lib/asterisk/sounds
and it stills request the sound file , and it give me
this message :
Use of uninitialized value in string ne at
/var/lib/asterisk/agi-bin/callback.agi line 56,
STDIN line 14.

do you have any idea what do i have to change in the
script? 





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[Asterisk-Users] Warning LSP Low

2005-11-23 Thread jonny hashem
Hi list:
what does this message mean Warning LSP Low and does
it relate to the bandwidth?




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[Asterisk-Users] Does Voipjet uses IAX2 trunking

2005-11-22 Thread jonny hashem
Hi list:
May be my question is silly ,but i want to know if
Voipjet uses IAX2 trunking becuase i have a bandwidth
of 64kbps dedicated and i can just make 2 calls in the
same time and if i the calls exceeds than 2 calls  it
breaking out voice and bad quality. 
 



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[Asterisk-Users] what is the role of trunk=yes

2005-11-09 Thread jonny hashem
Hi:
Can any body tell me what is the role of the trunk=yes
property that exists in iax.conf?
  
Regards;
jonny



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[Asterisk-Users] iax invtation problem

2005-10-16 Thread jonny hashem
i had a sip invitation problem with my voip provider
and here the message that was shown :
Oct 16 20:23:19 WARNING[21901]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to 'asterisk
sip:[EMAIL PROTECTED];tag=as7b43dfbd'
-- SIP/callshop-3fcc is circuit-busy
  == Everyone is busy/congested at this time
-- Got SIP response 481 Call Leg Does Not Exist
back from 213.61.187.150

My sip box is :195.112.214.99
The voip provider sip box:213.61.187.150

the configuration of my sip file was like this:

  sip.conf  

[callshop]
type=peer
host=213.61.187.150
username=X
secret=X

but when ive added these lines on my sip.conf file:
[callshop]
type=peer
host=213.61.187.150
username=X
secret=X
fromuser=
usereqphone=yes
canreinvite=no
nat=yes
insecure=invite
insecure=port
port=5060
disallow=all
allow=g729

it worked and the invitation problem was solved,but
when i have tried to send calls by iax to the same
voip provider the call failed like this:

dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp,
IAX2/callshopcompany/0017046872001) in new stack
-- Called callshopcompany/0017046872001
-- Call accepted by 213.61.187.150 (format g729)
-- Format for call is g729
-- Hungup 'IAX2/callshopcompany/3'
  == No one is available to answer at this time
Oct 16 20:35:53 WARNING[30499]: pbx.c:1949
ast_pbx_run: Timeout, but no rule 't' in context
'calls'
  Hangup on console 

My iax.conf file is like this:

[callshopcompany]
type=peer
host=sip.callshopcompany.com
username=875630553
secret=darWish472

the voip provider that iam dealing with accept both
sip and iax,but i realize in this case that something
must be added to the iax.conf file to solve the
invitation problem like i have done this in sip.conf
file.

Regards;
Jonny

  




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[Asterisk-Users] does tellular cell phones support answer switching

2005-10-10 Thread jonny hashem
i have tellular cell phone plugged to fxo modules ,the
problem that i am facing is when i dial a number on
the fxo modules the call is been answered before it
picked up on the other side , i thought that the
analoge lines do not support the answer switching
feature but not tellular cell phones because i think
it digital.




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[Asterisk-Users] iax invitation problem

2005-10-02 Thread jonny hashem
i have opened an account with  callshopcompany,and
when ive tried to send calls by the sip i had a
message show an asterisk invitation problem  i had
these sip configuration:
   sip.conf
[callshop]
type=peer
host=sip.callshopcompany.com
username=X
secret=XXX
 
Then i tried to add these lines and it worked :
   sip.conf

[callshop]
type=peer
host=sip.callshopcompany.com
username=XX
secret=XX
fromuser=XX
usereqphone=yes
canreinvite=no
nat=yes
insecure=invite
insecure=port
port=5060

but when i tried to send calls using iax the call
authenticated and then failed giving this:

-- Executing Dial(OSS/dsp,
IAX2/call/0017046872001) in new stack
-- Called call/0017046872001
-- Call accepted by 213.61.187.150 (format g729)
-- Format for call is g729
-- Hungup 'IAX2/call/2'
  == No one is available to answer at this time
Oct  2 23:32:37 WARNING[18702]: pbx.c:1949
ast_pbx_run: Timeout, but no rule 't' in context
'calls'
  Hangup on console 

The iax.conf file is like this:

[call]
type=peer
username=875630553
secret=darWish472
host=sip.callshopcompany.com






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[Asterisk-Users] asterisk invitation problem

2005-09-23 Thread jonny hashem
when i send calls from an asterisk box to a voip
provider the call fails and give me these messages:

*CLI Sep 23 21:32:32 WARNING[14595]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to 'asterisk
sip:[EMAIL PROTECTED]:5070;tag=as19e688a1'
-- SIP/call-0f60 is circuit-busy
  == Everyone is busy/congested at this time

but when ive tried to call using xlite softphone the
calls passed without any problems.

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[Asterisk-Users] astcc hangup problem

2005-08-30 Thread jonny hashem
does anyone had an experience with not hanging up the
call in astcc, thats my problem sometimes call does
not hang up automatically and even when pressing star
the call continue without stopping.
 
Regards;
jonny   




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[Asterisk-Users] iax and zap interface problem

2005-08-23 Thread jonny hashem
i have 2 iax servers one with analoge line plug into
the TDM card and the second server is do dial from,i
have made the connection between the 2 servers
sucessfuly and when it goes to TDM it dials the number
and the call is done ,but in bad and noisy sound
quality and with an echo. does anyone has an answer.

Regards;
jonny 

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[Asterisk-Users] Does intel 865 board works fine with Asterisk

2005-08-17 Thread jonny hashem
Hi:

I would like to know what are the issues I need to
look for in a chipset board so I can make sure it
works fine with digium cards and Asterisk . Is intel
board 865 fits the description?

Regards;


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[Asterisk-Users] h323 registration problem

2005-08-15 Thread jonny hashem
iam new to h323 ,i just download it and compile and it
gives me these messages on console :

Sep 13 00:20:30 WARNING[15443]: chan_oh323.c:4014
oh323_gk_check: Gatekeeper discovery failed.
-- Retrying gatekeeper registration.

does any body knows whats going on.




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[Asterisk-Users] Best Voip provider

2005-08-13 Thread jonny hashem
what is the best voip provider that provides good
service ,good voice quality and good rates . any one
have  an experience with voip providers advice me.

Regards;
jonny  




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[Asterisk-Users] Sip ports

2005-08-11 Thread jonny hashem
i have added port=5060 to sip client configuration but
it seems the same problem and in the same errors:

Aug 11 10:29:18 WARNING[9869]: chan_sip.c:843
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for
seqno 102 (Non-critical Response)


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[Asterisk-Users] is there cdrs for sip

2005-08-11 Thread jonny hashem
i have used astcc to open accounts to clients but now
i dont want to use astcc and i want to use sip cdrs.
 

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[Asterisk-Users] does SIP works behind the NAT

2005-08-10 Thread jonny hashem
i tried to connect 2 iax servers and it worked well
but when i tried to connect 2 Sip servers in the iax
configuration (one behind the NAT and the other with
real IP) it failed and give me this:

Aug 10 23:25:47 WARNING[28013]: chan_sip.c:843
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for
seqno 102 (Non-critical Response)

i think it is a NAT problem but i want to know how to
fix it.

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[Asterisk-Users] channel_pvt.h not found

2005-08-09 Thread jonny hashem
after editing the Make file i tried to compile
asterisk-oh323-0.6.5 but many errors displayed like
this:

[EMAIL PROTECTED] 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/ali/asterisk-oh323-0.6.5/wrapper'
./check_ver /home/ali/pwlib pwlib
./check_ver /home/ali/openh323 openh323
ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o
wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o
wrapgkserver.o
make[1]: Leaving directory
`/home/ali/asterisk-oh323-0.6.5/wrapper'
make[1]: Entering directory
`/home/ali/asterisk-oh323-0.6.5/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations
-D_REENTRANT -D_GNU_SOURCE
-I/root/src/asterisk/include -I../wrapper -g -c -o
chan_oh323.o chan_oh323.c
chan_oh323.c:37:34: asterisk/channel_pvt.h: No such
file or directory
chan_oh323.c: In function `oh323_exception':
chan_oh323.c:1145: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_indicate':
chan_oh323.c:1326: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_digit':
chan_oh323.c:1388: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_text':
chan_oh323.c:1410: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_call':
chan_oh323.c:1434: error: dereferencing pointer to
incomplete type
chan_oh323.c:1453: error: structure has no member
named `callerid'
chan_oh323.c:1455: error: structure has no member
named `callerid'
chan_oh323.c:1457: error: structure has no member
named `callerid'
chan_oh323.c: In function `oh323_hangup':
chan_oh323.c:1613: error: dereferencing pointer to
incomplete type
chan_oh323.c:1721: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_read':
chan_oh323.c:1749: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_write':
chan_oh323.c:2050: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_answer':
chan_oh323.c:2242: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_fixup':
chan_oh323.c:2286: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `ast_oh323_new':
chan_oh323.c:2518: error: dereferencing pointer to
incomplete type
chan_oh323.c:2527: error: dereferencing pointer to
incomplete type
chan_oh323.c:2529: error: dereferencing pointer to
incomplete type
chan_oh323.c:2536: error: dereferencing pointer to
incomplete type
chan_oh323.c:2537: error: dereferencing pointer to
incomplete type
chan_oh323.c:2538: error: dereferencing pointer to
incomplete type
chan_oh323.c:2539: error: dereferencing pointer to
incomplete type
chan_oh323.c:2540: error: dereferencing pointer to
incomplete type
chan_oh323.c:2541: error: dereferencing pointer to
incomplete type
chan_oh323.c:2542: error: dereferencing pointer to
incomplete type
chan_oh323.c:2543: error: dereferencing pointer to
incomplete type
chan_oh323.c:2544: error: dereferencing pointer to
incomplete type
chan_oh323.c:2545: error: dereferencing pointer to
incomplete type
chan_oh323.c:2546: error: dereferencing pointer to
incomplete type
chan_oh323.c:2547: error: dereferencing pointer to
incomplete type
chan_oh323.c:2548: error: dereferencing pointer to
incomplete type
chan_oh323.c:2549: error: dereferencing pointer to
incomplete type
chan_oh323.c:2550: error: dereferencing pointer to
incomplete type
chan_oh323.c:2551: error: dereferencing pointer to
incomplete type
chan_oh323.c:2552: error: dereferencing pointer to
incomplete type
chan_oh323.c:2579: error: structure has no member
named `dnid'
chan_oh323.c:2589: error: structure has no member
named `callerid'
chan_oh323.c:2590: error: structure has no member
named `callerid'
chan_oh323.c:2591: error: structure has no member
named `callerid'
chan_oh323.c:2596: error: structure has no member
named `callerid'
chan_oh323.c:2597: error: structure has no member
named `callerid'
chan_oh323.c:2598: error: structure has no member
named `callerid'
chan_oh323.c:2600: error: structure has no member
named `callerid'
chan_oh323.c:2605: error: structure has no member
named `callerid'
chan_oh323.c:2606: error: structure has no member
named `callerid'
chan_oh323.c:2608: error: structure has no member
named `callerid'
chan_oh323.c:2610: error: structure has no member
named `callerid'
chan_oh323.c:2614: error: structure has no member
named `callerid'
chan_oh323.c:2617: error: structure has no member
named `ani'
chan_oh323.c:2617: error: structure has no member
named `callerid'
chan_oh323.c:2623: error: structure has no member
named `callerid'
chan_oh323.c:2624: error: structure has no member
named `callerid'
chan_oh323.c: In function `oh323_request':
chan_oh323.c:2741: error: dereferencing pointer to
incomplete type
chan_oh323.c:2743: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_atexit':
chan_oh323.c:4923: 

[Asterisk-Users] low sound

2005-08-06 Thread jonny hashem
my customers complain that when they make a call they
hear the another side very well but the another side
hears the first side well but in low sound.what is the
ptoblem here and i have to change? 

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[Asterisk-Users] how may channels

2005-08-05 Thread jonny hashem
how many channels using codec g729 can be done by an
internet bandwidth to 512kb dedicated.

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[Asterisk-Users] how may channels

2005-08-05 Thread jonny hashem
how many channels using codec g729 can be used by an
internet bandwidth to 512kb dedicated.

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[Asterisk-Users] erros while updating the latest CVS

2005-07-27 Thread jonny hashem
Hello,

I just checked out the latest CVS and compiled and now
get the following error:

asterisk: symbol lookup error:
/usr/lib/asterisk/modules/res_config_mysql.so:
undefined symbol: ast_config_load


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[Asterisk-Users] problems with compiling asterisk-oh323-0.6.5

2005-07-26 Thread jonny hashem
after editing the Makefile according to my directories
ive compiled asterisk-oh323-0.6.5 but i got these
errors can any body help me with this :

[EMAIL PROTECTED]@net 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/wassim/asterisk-oh323-0.6.5/wrapper'
./check_ver /home/wassim/pwlib pwlib
./check_ver /home/wassim/openh323 openh323
ar rc liboh323wrap_s.a wrapper_misc.o asteriskaudio.o
wrapconnection.o wrapendpoint.o wrapper.o wrapcaps.o
wrapgkserver.o
make[1]: Leaving directory
`/home/wassim/asterisk-oh323-0.6.5/wrapper'
make[1]: Entering directory
`/home/wassim/asterisk-oh323-0.6.5/asterisk-driver'
gcc -Wall -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations
-D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include
-I../wrapper -g -c -o chan_oh323.o chan_oh323.c
chan_oh323.c:37:34: asterisk/channel_pvt.h: No such
file or directory
chan_oh323.c: In function `oh323_exception':
chan_oh323.c:1145: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_indicate':
chan_oh323.c:1326: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_digit':
chan_oh323.c:1388: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_text':
chan_oh323.c:1410: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_call':
chan_oh323.c:1434: error: dereferencing pointer to
incomplete type
chan_oh323.c:1453: error: structure has no member
named `callerid'
chan_oh323.c:1455: error: structure has no member
named `callerid'
chan_oh323.c:1457: error: structure has no member
named `callerid'
chan_oh323.c: In function `oh323_hangup':
chan_oh323.c:1613: error: dereferencing pointer to
incomplete type
chan_oh323.c:1721: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_read':
chan_oh323.c:1749: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_write':
chan_oh323.c:2050: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_answer':
chan_oh323.c:2242: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `oh323_fixup':
chan_oh323.c:2286: error: dereferencing pointer to
incomplete type
chan_oh323.c: In function `ast_oh323_new':
chan_oh323.c:2518: error: dereferencing pointer to
incomplete type
chan_oh323.c:2527: error: dereferencing pointer to
incomplete type
chan_oh323.c:2529: error: dereferencing pointer to
incomplete type
chan_oh323.c:2536: error: dereferencing pointer to
incomplete type
chan_oh323.c:2537: error: dereferencing pointer to
incomplete type
chan_oh323.c:2538: error: dereferencing pointer to
incomplete type
chan_oh323.c:2539: error: dereferencing pointer to
incomplete type
chan_oh323.c:2540: error: dereferencing pointer to
incomplete type
chan_oh323.c:2541: error: dereferencing pointer to
incomplete type
chan_oh323.c:2542: error: dereferencing pointer to
incomplete type
chan_oh323.c:2543: error: dereferencing pointer to
incomplete type
chan_oh323.c:2544: error: dereferencing pointer to
incomplete type
chan_oh323.c:2545: error: dereferencing pointer to
incomplete type
chan_oh323.c:2546: error: dereferencing pointer to
incomplete type
chan_oh323.c:2547: error: dereferencing pointer to
incomplete type
chan_oh323.c:2548: error: dereferencing pointer to
incomplete type
chan_oh323.c:2549: error: dereferencing pointer to
incomplete type
chan_oh323.c:2550: error: dereferencing pointer to
incomplete type
chan_oh323.c:2551: error: dereferencing pointer to
incomplete type
chan_oh323.c:2552: error: dereferencing pointer to
incomplete type
chan_oh323.c:2579: error: structure has no member
named `dnid'
chan_oh323.c:2589: error: structure has no member
named `callerid'
chan_oh323.c:2590: error: structure has no member
named `callerid'
chan_oh323.c:2591: error: structure has no member
named `callerid'
chan_oh323.c:2596: error: structure has no member
named `callerid'
chan_oh323.c:2597: error: structure has no member
named `callerid'
chan_oh323.c:2598: error: structure has no member
named `callerid'
chan_oh323.c:2600: error: structure has no member
named `callerid'
chan_oh323.c:2605: error: structure has no member
named `callerid'
chan_oh323.c:2606: error: structure has no member
named `callerid'
chan_oh323.c:2608: error: structure has no member
named `callerid'
chan_oh323.c:2610: error: structure has no member
named `callerid'
chan_oh323.c:2614: error: structure has no member
named `callerid'
chan_oh323.c:2617: error: structure has no member
named `ani'
chan_oh323.c:2617: error: structure has no member
named `callerid'
chan_oh323.c:2623: error: structure has no member
named `callerid'
chan_oh323.c:2624: error: structure has no member
named `callerid'
chan_oh323.c: In function `oh323_request':
chan_oh323.c:2741: error: dereferencing pointer to
incomplete type
chan_oh323.c:2743: error: dereferencing pointer to
incomplete type

[Asterisk-Users] does astcc support h323

2005-07-24 Thread jonny hashem
does astcc support h323 because i didnt find in the
trunks any technology named by h323.

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[Asterisk-Users] problems with tdm11p

2005-07-21 Thread jonny hashem
sometimes my tdm11p reads the caller id and sometimes 
doesnt read it and give me this :

Jul 21 13:55:50 NOTICE[6284]: callerid.c:307
callerid_feed: Caller*ID failed checksum
Jul 21 13:55:54 WARNING[6284]: chan_zap.c:5739
ss_thread: CallerID returned with error on channel
'Zap/4-1'

what is the problem please help.

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[Asterisk-Users] problems with tdm11P

2005-07-15 Thread jonny hashem
when i display in extensions.conf this :

exten = s,1,Answer
exten = s,n,DeadAGI(astcc.agi,${CALLERIDNUM})
exten = s,n,hangup

when a call comes the zap doesnot rerad the callerid
an give me this:

Jul 15 10:45:24 WARNING[6693]: chan_zap.c:5739
ss_thread: CallerID returned with error on channel
'Zap/4-1'

excuse me for my english iam italian ,thanks in
advance.






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[Asterisk-Users] bandwidth of gsm and g729

2005-07-14 Thread jonny hashem
what are the bandwidths of the gsm codec and g729
codec
and are they in same sound quality .




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[Asterisk-Users] bandwidth og gsm and g729

2005-07-14 Thread jonny hashem
what are the bandwidths of the gsm codec and g729
codec
and are they in same sound quality .

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[Asterisk-Users] how many g729

2005-07-13 Thread jonny hashem
how many g729 i need if i have 2 channels one is fxo
and the other is fxs,and how to display the 2 g729 on
iax .conf:
;allow=all  
disallow=all
;allow=ulaw  
allow=g729 

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[Asterisk-Users] choosing a softphone

2005-07-12 Thread jonny hashem
which the best softphone that works with window and
linux supporting IAX2 ,thanks in advance.



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[Asterisk-Users] choosing a softphone

2005-07-12 Thread jonny hashem
which is the best softphone that works with window and
linux supporting IAX2 ,thanks in advance.




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