[asterisk-users] cannot allocate memory

2009-02-25 Thread wassim Darwish

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 1GB of RAM  sufficient for this volume of calls on Asterisk.

 

and when iam using top command this is what i get:

 

top - 20:20:25 up 2:15, 1 user, load average: 0.57, 0.22, 0.13 
Tasks: 100 total, 1 running, 36 sleeping, 0 stopped, 63 zombie 
Cpu(s): 7.7% us, 2.3% sy, 0.0% ni, 90.0% id, 0.0% wa, 0.0% hi, 0.0% si, 
Mem: 1048576k total, 316088k used, 732488k free, 0k buffers 
Swap: 0k total, 0k used, 0k free, 0k cached

 

As i see it that the free memory is 732488k ,so it should'nt make this error.

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Re: [asterisk-users] linksys PAP2t and asterisk

2009-02-15 Thread wassim Darwish

Hi:
yes i think this is it ,but what is it and how can i remove it ? 



Date: Sat, 14 Feb 2009 14:23:27 -0700From: floj...@gmail.comto: 
asterisk-us...@lists.digium.comsubject: Re: [asterisk-users] linksys PAP2t and 
asteriskMan, as the CLI says:

SIP/us-092acb78 is ringing  (here it gives me 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 
provider:-- Executing [88017736288...@direct:1] Dial(SIP/490115-092bacc8, 
SIP/us/88017736288155) in new stack-- Called us/88017736288155-- Call 
on SIP/us-092acb78 left from hold-- SIP/us-092acb78 is making progress 
passing it to SIP/490115-092bacc8-- SIP/us-092acb78 is ringing  (here it 
gives me a fake ring) how can i disable this ringing . 


From: wassim...@hotmail.comto: asterisk-us...@lists.digium.comdate: Fri, 13 Feb 
2009 20:08:20 +Subject: [asterisk-users] linksys PAP2t and asterisk


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. 

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Re: [asterisk-users] linksys PAP2t and asterisk

2009-02-14 Thread wassim Darwish

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 us/88017736288155-- Call 
on SIP/us-092acb78 left from hold-- SIP/us-092acb78 is making progress 
passing it to SIP/490115-092bacc8-- SIP/us-092acb78 is ringing  (here it 
gives me a fake ring)
 
how can i disable this ringing . 



From: wassim...@hotmail.comto: asterisk-us...@lists.digium.comdate: Fri, 13 Feb 
2009 20:08:20 +Subject: [asterisk-users] linksys PAP2t and asterisk

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. 



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[asterisk-users] linksys PAP2t and asterisk

2009-02-13 Thread wassim Darwish

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.
 

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[asterisk-users] fake ring again when using SIP

2009-02-03 Thread wassim Darwish

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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Re: [asterisk-users] fake ringback tone

2009-01-10 Thread wassim Darwish

Hi:
Iam not using the 'r' option in my dial plan ,here what i have in my dial plan:
 
[gw]exten = _70.,1,Dial,SIP/grands/${EXTEN} Date: Fri, 9 Jan 2009 16:25:41 
-0500 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com 
Subject: Re: [asterisk-users] fake ringback tone  On Fri, 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 are using the r 
option in your Dial statement, remove it. That generates fake ringing. In 
FreePBX, that option is under the General settings, if plain jane Asterisk, 
just remove the r in your dial line.  --  Thanks, Steve Totaro 
+18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype)  
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[asterisk-users] fake ringback tone

2009-01-09 Thread wassim Darwish

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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[asterisk-users] digium cards with sangoma cards

2008-05-24 Thread wassim darwish

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 ?   

Thanks in advance;
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[asterisk-users] G723 on asterisk 1.4.1

2008-03-22 Thread wassim darwish

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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[asterisk-users] polarity in zapata.conf

2008-03-20 Thread wassim darwish

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 want to do is to not let polarity reversal take effect on fxo number 4.
that what i have in my zapata.conf:

answeronpolarityswitch=yes
hanguponpolarityswitch=yes
signalling=fxs_ks
context=wassim
channel = 1-3
;
signalling=fxs_ks
context=wassim
channel = 4

Thanks in advance;



 
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[asterisk-users] FXO modules and polarity reverse

2008-02-06 Thread wassim darwish

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
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Re: [asterisk-users] Realtime Mysql error

2007-10-29 Thread wassim darwish

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 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 console and that what 
i've got: [Oct 27 01:05:32] ERROR[2607]: res_config_mysql.c:637 
mysql_reconnect: MySQL RealTime: Ping failed (2006). Trying an explicit 
reconnect. Connected to [EMAIL PROTECTED], port 3306 with username root for 9 
hours, 43 minutes, 39 seconds. Can any body help with this; Hi what is the 
version of asterisk and mysql what distro you are using ram

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[asterisk-users] Realtime Mysql error

2007-10-26 Thread wassim darwish

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 console and 
that what i've  got:
[Oct 27 01:05:32] ERROR[2607]: res_config_mysql.c:637 mysql_reconnect: MySQL 
RealTime: Ping failed (2006).  Trying an explicit reconnect.
Connected to [EMAIL PROTECTED], port 3306 with username root for 9 hours, 43 
minutes, 39 seconds.

Can any body help with this;

Thanks in advance;
Wassim




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[asterisk-users] Astcc does'nt count all calls

2007-09-30 Thread wassim darwish

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 possible that this problem occured due to launching mysql by this command 
:
 /etc/init.d/mysqld start --force 

Can any body help me please.

Thanks in advance;
wassim
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[asterisk-users] astcc sometimes doesnt write on mysql

2007-09-27 Thread wassim darwish

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;
wassim
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Re: [asterisk-users] Musiconhold instead ringing

2007-09-08 Thread wassim darwish

 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 want 
asterisk to make the caller hear musicmusiconhold instead of ringing,Can 
any body help me with this? my guess is you'd have to Answer() the call 
first, then play moh while Dial()'ing the exten. -- Anthony - 
http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 
BCF0 10BE 9967 92DC 35DC B001 4A4E

Hi:
Can you represent it in extensions Dialplan ?

Regards;
Wassim
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Re: [asterisk-users] Musiconhold instead ringing

2007-09-08 Thread wassim darwish

 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: 
 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 want asterisk to make the caller hear musicmusiconhold 
instead of ringing,Can any body help me with this? my guess is you'd have 
to Answer() the call first, then play moh while Dial()'ing the exten. -- 
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 
5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E Hi: Can you represent it 
in extensions Dialplan ? Untested, but something like this should work 
exten = _X.,1,Answer extem = 
_X.,2,Dial(SIP/${EXTEN}|50|m(musiconhold-class) You have to fix this to match 
Your existing dialplan (Extensions, SIP-Accounts...). It won't work without 
the answer-statement. HTH, Karsten 
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Hi:
Thank you all for your suggestions,it worked well using Karsten suggestion.

Thanks again,
Wassim

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[asterisk-users] Musiconhold instead ringing

2007-09-07 Thread 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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[asterisk-users] No Dial tone came from fxs modules

2007-09-05 Thread wassim darwish

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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Re: [asterisk-users] No Dial tone came from fxs modules

2007-09-05 Thread wassim darwish

 Date: Wed, 5 Sep 2007 09:21:19 -0800 
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: 
[asterisk-users] No Dial tone came from fxs modules Just to be clear, I 
thought that dialtone provision didn't require the power cable, just 
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 it . Can i get some help please. do you have the power cable 
attached to it. that's what you need to generate a dialtone. 
 
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Hi:
I checked the power cable  and its plugged in the TDM, Is there anything else 
to check?
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Re: [asterisk-users] Zaptel modules are being installed in different directory

2007-09-02 Thread wassim darwish

 Date: Sun, 2 Sep 2007 04:31:22 -0500 
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: 
[asterisk-users] Zaptel modules are being installed in different directory 
Tzafrir Cohen wrote: On Sat, Sep 01, 2007 at 01:23:47PM -0500, Eric 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 of: rpm -qa | grep kernel ls -l 
/lib/modules/`uname -r`/build/.config The easiest thing to do is 
remove the 2.6.8.1-12mdkcustom directory, create a symlink from 
2.6.8.1-12mdkcustom to 2.6.8.1-12mdk, and reinstall zaptel. Are you 
sure? I suspect a: modinfo ./zaptel.ko would show that the kernel 
version string there is 2.6.8.1-12mdkcustom and hence it won't load. I 
suspect that Steve's advice is generally correct. I just wonder if there's a 
more striahgt-forward way on Mandrake, or if they give a different kernel 
version on purpuse in case you decide to rebuild the kernel. You have not 
replied to my other questions (what kernel source / headers rpm packages are 
installed). Anybody here more familiar with Mandriva's current magics on this 
subject? You edit /usr/src/linux/Makefile and modify the EXTRAVERISON 
setting. Then you spend a while trying to get the zaptel build process to not 
keep installing the modules in the wrong directory, get frustrated, give up, 
rm -f the zaptel source, untar zaptel, build install. Then wish you had just 
done the symlink in the first place. 
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Hi:
Thank you all for your suggestions,i tried hard at mandrake but without any new 
results.
so i tried to install Fedora core 5 ,Every thing went well ,i did modprobe 
wctdm and the TDM activated properly but when i tried to open /etc/zaptel.conf 
to configure i didnt find it,There is no zaptel.conf in /etc .

I appreciate your help 
Thanks;
Wassim
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[asterisk-users] Zaptel modules are being installed in different directory

2007-09-01 Thread wassim darwish

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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Re: [asterisk-users] Zaptel modules are being installed in different directory

2007-09-01 Thread wassim darwish

 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: 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 -- Tzafrir Cohen icq#16849755 
jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] 
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir 
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Hi :
The output of  uname -r is 2.6.8.1-12mdk
The output of ls -l /lib/modules/2.6.8.1-12mdk/build/.config is  -rw-r--r--  
1 root root 60428 Oct  1  2004 /lib/modules/2.6.8.1-12mdk/build/.config


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Re: [asterisk-users] Zaptel modules are being installed in different directory

2007-09-01 Thread wassim darwish

 Date: Sat, 1 Sep 2007 13:23:47 -0500 
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: 
[asterisk-users] Zaptel modules are being installed in different directory 
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 of: rpm -qa | grep 
kernel ls -l /lib/modules/`uname -r`/build/.config The easiest thing to 
do is remove the 2.6.8.1-12mdkcustom directory, create a symlink from 
2.6.8.1-12mdkcustom to 2.6.8.1-12mdk, and reinstall zaptel. This is a common 
issue for Mandrake/Mandriva users. I'm sure there is a more correct fix, but I 
do the symlink thing and never have to worry about it. 
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Hi:
Iam new to linux world, Can you just explain how can i create symlink from 
2.6.8.1-12mdkcustom to 2.6.8.1-12mdk ?
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Re: [asterisk-users] Zaptel modules are being installed in different directory

2007-09-01 Thread wassim darwish

 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: Sat, 1 
Sep 2007 13:23:47 -0500 From: [EMAIL PROTECTED] To: 
asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Zaptel modules 
are being installed in different directory 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 of: rpm -qa | grep kernel ls -l /lib/modules/`uname 
-r`/build/.config 
The easiest thing to do is remove the 2.6.8.1-12mdkcustom directory, create a 
symlink from 2.6.8.1-12mdkcustom to 2.6.8.1-12mdk, and reinstall zaptel. 
This is a common issue for Mandrake/Mandriva users. I'm sure there is a more 
correct fix, but I do the symlink thing and never have to worry about it. 
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UNSUBSCRIBE or update options visit: 
http://lists.digium.com/mailman/listinfo/asterisk-users Hi: Iam new to 
linux world, Can you just explain how can i create symlink from 
2.6.8.1-12mdkcustom to 2.6.8.1-12mdk ? I think the more correct fix is to 
edit the name of the kernel and remove the custom before compiling it. I 
rename all of my custom kernels but never played with this in Mandrake. 
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Hi:
i have in my /lib/modules before installing zaptel-1.4.5 this:
[EMAIL PROTECTED] modules]# ls
2.6.8.1-12mdk/
but after installing zaptel-1.4.5  this :
[EMAIL PROTECTED] modules]# ls
2.6.8.1-12mdk/  2.6.8.1-12mdkcustom/

By the way when i have done installing zaptel-1.4.5 i tried to get to 
/etc/zaptel.conf but it seems not exists. 

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[Asterisk-Users] callback script

2005-12-02 Thread wassim darwish
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.

if any body knows about this configuration ,just show
me what to put in the meetme and enhanced-outgoing
contexts and what to edit in this part of callback.agi
script:

$outgoingclid = ;
$channel = ;
$context = ;
$church = ;

Regards;
wassim




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[Asterisk-Users] where can I find this property: answeronpolarityswitch

2005-10-02 Thread wassim darwish
where can i find these properties:
answeronpolarityswitch and hanguponpolarityswitch.

Regard;
wassim



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[Asterisk-Users] sip problem

2005-08-12 Thread wassim darwish
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 an asterisk ):

Aug 12 18:19:11 NOTICE[12149]: chan_sip.c:5284
register_verify: Peer 'wassim' is trying to register,
but not configured as host=dynamic
Aug 12 18:19:11 NOTICE[12149]: chan_sip.c:8730
handle_request_register: Registration from
'sip:[EMAIL PROTECTED]' failed for
'195.112.214.98'
Aug 12 18:19:11 WARNING[12149]: chan_sip.c:7910
handle_response: Forbidden - wrong password on
authentication for REGISTER for 'wassim' to
'195.112.214.98'
Aug 12 18:19:31 NOTICE[12149]: chan_sip.c:4380
sip_reg_timeout:-- Registration for
'[EMAIL PROTECTED]' timed out, trying again

any body have an idea.


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[Asterisk-Users] iax to iax severs

2005-08-03 Thread wassim darwish
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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[Asterisk-Users] most stable linux to build business

2005-07-28 Thread wassim darwish
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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[Asterisk-Users] how to compile asterisk-oh323

2005-07-26 Thread wassim darwish
if any one can tell how to compile asterisk-oh323 and
what it is dependencies.

Regards;
wassim 




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[Asterisk-Users] cannot find channel_pvt.h

2005-07-26 Thread wassim darwish
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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[Asterisk-Users] does h323 exists in astcc trunks

2005-07-25 Thread wassim darwish
does astcc support h323 ,because it doesnt exists h323
in trunks technology.

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[Asterisk-Users] problems with compiling asterisk-oh323

2005-07-25 Thread wassim darwish
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 COPYINGREADMErules.mak 
wrapper

asterisk-oh323-0.6.6]# make

for x in wrapper asterisk-driver; do make -C $x build
|| exit 1 ; done
make[1]: Entering directory
`/home/wassim/asterisk-oh323-0.6.6/wrapper'
./check_ver /root/src/oh323/pwlib pwlib
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'.  Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'.  Stop.
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
cat: /root/src/oh323/pwlib/version.h: No such file or
directory
./check_ver /root/src/oh323/openh323 openh323
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'.  Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'.  Stop.
cat: /root/src/oh323/openh323/version.h: No such file
or directory
cat: /root/src/oh323/openh323/version.h: No such file
or directory
cat: /root/src/oh323/openh323/version.h: No such file
or directory
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'.  Stop.
g++ -Wall -x c++ -Os -DUSE_OLD_CAPABILITIES_API=1  
-DWRAPTRACING -DWRAPTRACING_LEVEL=5
-DPWLIBVERSION=\..\ -DOPENH323VERSION=\..\ 
-I/root/src/oh323/pwlib/include
-I/root/src/oh323/openh323/include
-I/root/src/oh323/openh323/include/openh323
-I../asterisk-driver -c wrapper_misc.cxx -o
wrapper_misc.o
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'.  Stop.
openh323flags.mak:2:
/root/src/oh323/openh323/openh323u.mak: No such file
or directory
make[1]: *** No rule to make target
`/root/src/oh323/openh323/openh323u.mak'.  Stop.
In file included from wrapper_misc.cxx:34:
wrapper_misc.hxx:35:19: ptlib.h: No such file or
directory
In file included from wrapper_misc.cxx:34:
wrapper_misc.hxx:61: error: expected class-name before
'{' token
wrapper_misc.hxx:63: error: `PMutex' has not been
declared
wrapper_misc.hxx:63: error: ISO C++ forbids
declaration of `PCLASSINFO' with no type
wrapper_misc.hxx:63: error: ISO C++ forbids
declaration of `parameter' with no type
wrapper_misc.hxx:68: error: `BOOL' does not name a
type
wrapper_misc.hxx:73: error: `PString' does not name a
type
wrapper_misc.cxx: In constructor
`WrapMutex::WrapMutex(char*)':
wrapper_misc.cxx:48: error: class `WrapMutex' does not
have any field named `PMutex'
wrapper_misc.cxx:50: error: `name' undeclared (first
use this function)
wrapper_misc.cxx:50: error: (Each undeclared
identifier is reported only once for each function it
appears in.)
wrapper_misc.cxx:50: error: `PString' undeclared
(first use this function)
wrapper_misc.cxx:51: error: `cout' undeclared (first
use this function)
wrapper_misc.cxx:51: error: 'class WrapMutex' has no
member named 'Class'
wrapper_misc.cxx:51: error: `endl' undeclared (first
use this function)
wrapper_misc.cxx: At global scope:
wrapper_misc.cxx:54: error: `BOOL' does not name a
type
wrapper_misc.cxx: In member function `void
WrapMutex::Signal(const char*, int, const char*)':
wrapper_misc.cxx:78: error: `PMutex' has not been
declared
wrapper_misc.cxx:78: error: no matching function for
call to `WrapMutex::Signal()'
wrapper_misc.cxx:77: note: candidates are: void
WrapMutex::Signal(const char*, int, const char*)
wrapper_misc.cxx:79: error: `cout' undeclared (first
use this function)
wrapper_misc.cxx:79: error: 'class WrapMutex' has no
member named 'Class'
wrapper_misc.cxx:79: error: `name' undeclared (first
use this function)
wrapper_misc.cxx:79: error: `endl' undeclared (first
use this function)
make[1]: *** [wrapper_misc.o] Error 1
make[1]: Leaving directory
`/home/wassim/asterisk-oh323-0.6.6/wrapper'
make: *** [subdirs_build] Error 1




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[Asterisk-Users] G729 with 2 channels

2005-07-16 Thread wassim darwish
how to configure  the g729 with 2 channels in iax.conf.

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[Asterisk-Users] problems with g729

2005-07-14 Thread wassim darwish
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!







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[Asterisk-Users] asking again

2005-07-12 Thread wassim Darwish
ok what softphone i should use to fit windows and linux supporting 
iax,thanks in advance.


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[Asterisk-Users] gnophone installation

2005-07-11 Thread wassim Darwish
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.


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[Asterisk-Users] how to edit ring time

2005-07-09 Thread wassim darwish
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. 




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[Asterisk-Users] editing ring time

2005-07-08 Thread wassim darwish
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.

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[Asterisk-Users] changing Nobody picked up in 30000 m

2005-07-08 Thread wassim darwish
i dont know how to edit the the time for ringing
3ms to 4ms,please help me. 

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[Asterisk-Users] changing Nobody picked up in 30000 ms

2005-07-07 Thread wassim darwish
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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[Asterisk-Users] editing time in astcc

2005-07-03 Thread wassim darwish
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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[Asterisk-Users] raising the sound volume on zap

2005-07-03 Thread wassim darwish
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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[Asterisk-Users] editing time to say astcc-noanswer

2005-07-02 Thread wassim darwish
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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[Asterisk-Users] how does pattern routes works

2005-07-01 Thread wassim darwish
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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[Asterisk-Users] problems in dialing in routes patterns

2005-06-30 Thread wassim darwish
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.

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[Asterisk-Users] HOW TO WRITE ROUTE PATTERNS DIALPLAN

2005-06-28 Thread wassim darwish
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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[Asterisk-Users] facing troubles with routes patterns dialplan

2005-06-27 Thread wassim darwish
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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[Asterisk-Users] failure in writing in pattern (routes)

2005-06-26 Thread wassim darwish
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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[Asterisk-Users] facing problems with TDM400P

2005-06-04 Thread wassim darwish
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.




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[Asterisk-Users] dialling problem with astcc

2005-04-22 Thread wassim darwish
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 script or is it from the telephone
system
i dont know what to do please help. 

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[Asterisk-Users] delay problem in asterisk

2005-04-14 Thread wassim darwish
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 ,please help me i dont
know what to do.




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[Asterisk-Users] HOW TO SET THE TIME TO DIAL AFTER astcc-accountnum and astcc-phonenum

2005-04-09 Thread wassim darwish
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. 





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[Asterisk-Users] astcc problems

2005-04-02 Thread wassim darwish
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 Environment Dump:
 -- accountcode =
 -- callerid = 01475969
 -- calleridname = unknown
 -- channel = Zap/1-1
 -- context = incoming
 -- dnid = unknown
 -- enhanced = 0.0
 -- extension = s
 -- language = en
 -- priority = 3
 -- rdnis = unknown
 -- request = astcc.agi
 -- type = Zap
 -- uniqueid = 1112430048.4
Apr  2 03:20:54 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-tone does not exist in
any format
Res is
Silent Level is
Card no is 12345

Card has face value 3 and has used 0


3 dollars and 0 cents remain
Apr  2 03:20:54 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-youhave does not exist
in any format
-- Playing 'digits/3' (language 'en')
Apr  2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-dollars does not exist
in any format
Apr  2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-remaining does not
exist in any format
Apr  2 03:20:55 WARNING[11364]: file.c:486
ast_openstream_full: File astcc-badphone does not
exist in any format
-- AGI Script astcc.agi completed, returning 0

I dont know what the problem and what this warnings
mean and how can i fix them please help.
and thanks
 



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