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
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
Building a
Hi:
Iam looking for a sip softphone that supports g729 codec
Any one have an idea ?
Reagrds;
jonnyhashem
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Hi:
Is the difference of Asterisk verisons on two servers
effect on the iax2 trunking between them ?
Thanks;
jonny
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Hi:
What are the steps to make a trunked iax2 ?
Thanks in advance;
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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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HI:
what is command on console to know how many calls are
sending in the same time?
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--Bandwidth
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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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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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?
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
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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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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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
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
Hi list:
what does this message mean Warning LSP Low and does
it relate to the bandwidth?
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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.
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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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
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
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
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'
--
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
Start
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
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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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.
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
Start your day with Yahoo! - make it your home page
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)
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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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
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'
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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how many channels using codec g729 can be done by an
internet bandwidth to 512kb dedicated.
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how many channels using codec g729 can be used by an
internet bandwidth to 512kb dedicated.
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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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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 $x build
|| exit 1 ; done
make[1]: Entering directory
does astcc support h323 because i didnt find in the
trunks any technology named by h323.
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sometimes my tdm11p reads the caller id and sometimes
doesnt read it and give me this :
Jul 21 13:55:50 NOTICE[6284]: callerid.c:307
callerid_feed: Caller*ID failed checksum
Jul 21 13:55:54 WARNING[6284]: chan_zap.c:5739
ss_thread: CallerID returned with error on channel
'Zap/4-1'
what is the
when i display in extensions.conf this :
exten = s,1,Answer
exten = s,n,DeadAGI(astcc.agi,${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
what are the bandwidths of the gsm codec and g729
codec
and are they in same sound quality .
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what are the bandwidths of the gsm codec and g729
codec
and are they in same sound quality .
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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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which the best softphone that works with window and
linux supporting IAX2 ,thanks in advance.
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which is the best softphone that works with window and
linux supporting IAX2 ,thanks in advance.
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