Hi:
i have a hosted server with asterisk and a2billing as a billing plattform, when
i am trying to enter the server remotely by ssh, memory error message
displayed:
-bash: fork: Cannot allocate memory
i have 1GB RAM on the system ,and there is 15 to 25 concurrent calls on the
system is'nt
a fake ring)
It's the channel SIP/us/something, which is generating ring signalling.
2009/2/14 wassim Darwish wassim...@hotmail.com
this post is attached to the prevoius post, this is what i have on CLI when i
call from Linksys pap2t to asterisk and then asterisk bridge the call to a sip
this post is attached to the prevoius post, this is what i have on CLI when i
call from Linksys pap2t to asterisk and then asterisk bridge the call to a sip
provider:
-- Executing [88017736288...@direct:1] Dial(SIP/490115-092bacc8,
SIP/us/88017736288155) in new stack-- Called
Hi all:
when i make a call from linksys pap2t to an asterisk server a fake ring is
heard some times ,but when sending calls between 2 asterisk servers through sip
no fake ring is heard but real one.
any suggestions please.
_
Hi:
I cant figure it out why fake ring is heared when dialing through SIP
(Asterisk) ,it often it gives me fake ring but not always ,in some calls it
gives me real ring.
the dialplan is without 'r' option.
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, Jan 9, 2009 at 3:57
PM, wassim Darwish wassim...@hotmail.com wrote: hi: When iam sending
calls through sip a fake ringback tone is generated and then call status
can't be viewed (if call is ringing,busy,offline) it just rings and rings.
Can i disable this? Thanks in advance. If you
hi:
When iam sending calls through sip a fake ringback tone is generated and then
call status can't be viewed (if call is ringing,busy,offline) it just rings and
rings.
Can i disable this?
Thanks in advance.
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Hi:
Iam an Asterisk user and i have a Sangoma A200 with 4 fxo modules and i want
to buy Digium card with 4 fxo modules and insert it on the PCI besides the
sangoma card ,so i will have 8 fxo channels on my asterisk box ,Is that right?
Does Asterisk make errors if there is two different cards ?
Hi:
How to install and set up my asterisk server with G723 codec to send and
receive calls using it.
Thanks in advance;
Wassim
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hi:
In my zapata.conf i have 4 fxo configured channels,for fxo number 1 to 3 i
added polarity reversal property but for fxo number 4 i didnt add polarity
reversal property but it still giving me on cosole that fxo number 4 is
polarized (because the line on fxo number 4 is not polarized).
what
I would like to know if any body tried to connect gsm gateway with polarity
reversal to fxo module at asterisk server ,and if the polarity reversal solve
the problem of the answer and hangup supervison on calls .i appreciate any
help.Thanks in advance;Wassim
Hi:
Iam using Fedora core 5 .
Thanks in advance;
Date: Mon, 29 Oct 2007 10:23:28 +0530 From:
[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re:
[asterisk-users] Realtime Mysql error On 10/27/07, wassim darwish wrote:
Hi: Iam using an asterisk
Hi:
Iam using an asterisk server with astcc ,iam facing a problem with astcc that
when the call is hangup sometimes astcc doesnt calculate the call cost and the
call time and without writing the call status on cdrs table .
I tried to run this command realtime mysql status on the asterisk
Hi:
Iam using astcc on my asterisk server,sometimes astcc does'nt count calls by
not writing them into mysql ,example: Not subtracting the call cost from the
face value of the entered card number and by not writing it into cdrs
table.This problem occured sometimes and not always.
Is it
Hi:
I noticed that astcc on my asterisk server sometimes it doesnt write on mysql
,example :when the caller hangup the call its didnt written on cdrs table nor
subtract the cost of the call from the face value of caller card number.This
problem occured sometimes and not always.
Regards;
From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Date: Fri, 7 Sep 2007 19:10:04 -0500 Subject:
Re: [asterisk-users] Musiconhold instead ringing On Friday 07 September 2007
07:02:01 pm wassim darwish wrote: Hi: When i get an incoming call, i
From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Date: Sat, 8 Sep 2007 14:48:18 +0200 Subject:
Re: [asterisk-users] Musiconhold instead ringing Hi, Am Samstag, den
08.09.2007, 09:44 + schrieb wassim darwish
Hi:
When i get an incoming call, i want asterisk to make the caller hear
musicmusiconhold instead of ringing,Can any body help me with this?
Best regards;
Wassim
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Hi:
I have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made
modprobe wctdm the fxs modules is lightened but there is no dial tone came from
it .
Can i get some help please.
Best Regards;
Wassim
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generating ring voltages? Can anyone say? Moj Anthony Messina wrote: On
Wednesday 05 September 2007 09:09:25 am wassim darwish wrote: Hi: I
have asterisk-1.4.0 with zaptel 1.4.0 and TDM40P on my server ,when i made
modprobe wctdm the fxs modules is lightened but there is no dial tone came
from
ManxPower
Wieling wrote: Tzafrir Cohen wrote: On Sat, Sep 01, 2007 at 07:30:37AM
+, wassim darwish wrote: Hi: Iam running kernel is 2.6.8.1-12mdk
but the modules of zaptel are being installed to
/lib/modules/2.6.8.1-12mdkcustom how can i fix this up, any one have an
idea? What is the output
Hi:
Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel are being
installed to /lib/modules/2.6.8.1-12mdkcustom
how can i fix this up, any one have an idea?
Best Regards;
Wissam
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Date: Sat, 1 Sep 2007 11:55:28 +0300
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re:
[asterisk-users] Zaptel modules are being installed in different directory On
Sat, Sep 01, 2007 at 07:30:37AM +, wassim darwish wrote: Hi
, wassim darwish
wrote: Hi: Iam running kernel is 2.6.8.1-12mdk but the modules of zaptel
are being installed to /lib/modules/2.6.8.1-12mdkcustom how can i fix
this up, any one have an idea? What is the output of: rpm -qa | grep
kernel ls -l /lib/modules/`uname -r`/build/.config The easiest thing
Date: Sat, 1 Sep 2007 16:03:46 -0400
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re:
[asterisk-users] Zaptel modules are being installed in different directory
wassim darwish wrote: Date
Hi:
Once i have seen the post of Darren Wiebe of
suggestion of a callback configuration in
extensions.conf and it was like this:
[callback]
exten =
_.,1,AGI(callback.agi,LAKEVIEW,1234567890,9998,,meetme,enhanced-outgoing)
But i didnt know what to add in meetme and
enhanced-outgoing contexts.
where can i find these properties:
answeronpolarityswitch and hanguponpolarityswitch.
Regard;
wassim
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--Bandwidth and
i have configured a sip phone to make calls through a
sip server but when i make call through the sip phone
to the sip server every thing goes well and the call
is done perfectly but on sip server it gives me these
messages(i have 2 pc with different ips one with a sip
phone and the another with
if any one can tell me how to configure the iax files
of 2 iax servers ,one behind the NAT and the other
real ip
and the one behind the NAT requesting the other with
real ip.
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what is the most stable linux that we can build
business on it, i mean the best linux a linux without
problems .
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if any one can tell how to compile asterisk-oh323 and
what it is dependencies.
Regards;
wassim
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
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when i tried to compile asterisk-oh323 i get an error
that channel_pvt.h is missing,where i can find and
download it and in which directory i must put it.
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does astcc support h323 ,because it doesnt exists h323
in trunks technology.
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i ve downloaded
asterisk-oh323-0.6.6.tar.gz
I am getting this and anybody know howto fix this?
#tar zxvf asterisk-oh323-0.6.6.tar.gz
oh323]# cd asterisk-oh323-0.6.6
asterisk-oh323-0.6.6]# ls
asterisk-driver CONFIGURATION Makefile rpm
TESTS
BUGS COPYINGREADME
how to configure the g729 with 2 channels in iax.conf.
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when i display g729 on iax.conf and make a call using
g729 it gives this in several lines:
Jul 14 17:30:58 WARNING[14196]: codec_g729.c:180
g729tolin_framein: Out of G.729 Decoder Licenses!
Start your day with Yahoo! -
ok what softphone i should use to fit windows and linux supporting
iax,thanks in advance.
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i ve tried to find a gnophone dependency libgtksuperwin.so i searched
every where in google in wiki pages but i didnt found it at all ,if any one
can help in finding it i ll be thankful,and thanks in advance.
_
Dont just
i dont how to edit the time for ringing 3ms to
4ms when it displayed on console Nobody picked
up in 3 ms and its very short time for ringing .
please if anyone can help me do it please.
Sell on Yahoo! Auctions
how to edit the time of ring 3ms to 4ms in
astcc since it displays this on console Nobody picked
up in 3 ms when nobody picked up the phone in
3ms and then it hangup.
please help i have been asking this question from long
time and no body answered me yet.
i dont know how to edit the the time for ringing
3ms to 4ms,please help me.
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how to edit the time 3 ms for ringing to 4
ms, i ve tried but i dindt know how,so please help me please.
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i dont know how to edit the time 3ms for ringing
in astcc when it says there is no body to answer.i
want to change this time to 4ms but i dont know
how.please help please.
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i noticed that the sound volume of the zap(tdm400p)
was low ,so i tried to raise the sound volume but i
didnt know how please help me.
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i dont know how to edit the time 3ms for ringing
in astcc when it says there is no body to answer.i
want to change this time to 4ms but i dont know
how.please help please.
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i tried to write to usa destination 1* it worked well
but when i tried to specify the number of digits i
wrote
1NXXNXX but it did'nt work.can anybody help me
please
please.
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i tried to write a pattern to usa destination and that
was 1* it worked well but when i tried to specify the
number of digits i wrote 1NXXNXX it didnt work.
so what i must to write in case to specify the number
of digits of the destination.
__
in routes pattern i tried to write pattern to usa
destination and that was 1* it worked well but when i
wanted to specify the number of digits then i tried
1NXXNXX but i didnt work.so i dont know what to
write please help.
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i tried to write in routes, patterns to usa
destination 1.* it worked well but i wanted to specify
the number of digits then i tried 1NXXNXX but it
didnt work.
please help
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hi,i tried to write in pattern in routes to usa
destination 1* but i want to specify the number of
digits so i tried 1NXXNXX but it dose'nt worked so
please help me.
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i pulgged the TDM400P to my computer and when turning
on and dialing on zap the following message is shown :
app_dial.c:960 dial_exec_full: Unable to create
channel of type 'Zap' (cause 0)
== Everyone is busy/congested at this time (1:0/0/1)
please help me thanks lot.
when a call comes on zap astcc.agi script launch and
ask
caller about his card number,and when the caller is
dialing his card number(56170) sometimes astcc take it
by missing a number as (5670) or doubled number as
(556170)
i dont know whats the problem is it from zap or is it
from astcc.agi
i have asterisk on my system and when making a call a
delay problem in talking appears,that means when i
talk to somebody he will listen me after almost a
second (the ping on my voip provider's IP is 700ms to
800ms)so i dont know if the problem is in the nternet
connection or another problem
when a call comes the astcc-accountnum plays and ask
the caller about the card number and after playing
astcc-accountnum a period of time is given for the
caller to dial his card number but the problem here
is the short of the time given ,and i dont know where
and how can i setup the time.
i have downloaded astcc and confiugured it on web but
the problem is when a call comes by the right callerid
it gives me on CLI like this:
-- Executing DeadAGI(Zap/1-1,
astcc.agi|01475969|s) in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/astcc.agi
Detected dry run!
AGI
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