Hello,
How can I reset CDR time , let's say after 30 seconds of answer signal,
reset CDR to 0 , any idea ??
Thanks.
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Behalf Of Matt Riddell
Sent: Wednesday, September 09, 2009 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RESET CDR
On 9/09/09 4:34 PM, B.Masoud @ SH wrote:
> Hello,
>
> How can I reset CDR time , let's say after 30 seconds of answer s
ists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Wednesday, September 09, 2009 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RESET CDR
On 9/09/09 5:14 PM, B.Masoud @ SH wrote:
> A little more help is appreci
Discussion
Subject: Re: [asterisk-users] RESET CDR
On Wed, Sep 9, 2009 at 10:12 AM, B.Masoud @ SH wrote:
> I don't want to bill the first 30 seconds, that's all, why is that so
> strange??? My line does not support polarity reversal, so I don't want to
> bill for ringing the
Yes that is the problem.
So what do you do when the line doesn't support polarity??
What is the best solution in this case?
Thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
Sent: Thursday,
-Commercial Discussion
Subject: Re: [asterisk-users] RESET CDR
B.Masoud @ SH wrote:
> Yes that is the problem.
> So what do you do when the line doesn't support polarity??
> What is the best solution in this case?
What kind of gateway do you use to connect to the PSTN?
--
Iván Stepaniuk
Alb
Is there any program Asterisk users use to calculate ASR and ACD ??
Thanks for any comments.
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isk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Thursday, September 10, 2009 3:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ASR & ACD
Is there any program Asterisk users use to calculate ASR
I have added 2 trunk sequence in my outbound routes,
The problem is that:
1. If the call was busy on the first trunk it will go to the second
(i.e. the called party hung-up without answering the call)
How to overcome this???
___
-- Band
Hi
I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls
to that trunk, I am getting all circuits are busy now, do I have to do
something specific?? I am using elastix.
Thanks.
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ion (from-internal, 966505103250, 5) exited non-zero on
'SIP/100-08fba098'
-- Executing [...@from-internal:1] Macro("SIP/100-08fba098", "hangupcall")
in new stack
-- Executing [...@macro-hangupcall:1] ResetCDR("SIP/100-08fba098", "w") in
new s
Are you series???
My card is FXO TDM2400, I am sure its designed to forward calls to pstn!!!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira
Sent: Monday, October 05, 2009 5:07 AM
To: Asterisk Users Mailing
Man, thanks a lot!
I just changed the name to g0 instead of DGTDM24 and it worked!!
I would like to know where I can set the configuration for line tones( dial
tone, call and busy tone) and where I can change different setting for
polarity / current disconnect etc.. of the line?
I cant find Zapat
Thanks,
I made the zone, and now call disconnect works ok!
i have one last problem, I have defined the card g0 to have 24 channels, but
every time I try to call, if all ports are off the call always go to the
first port, how can I balance the calls over all ports???
Any suggestions appreciated.
Hello,
I would like to know how Asterisk deal in this case:
Assume I have a Main Asterisk Server located in UK, and another box that
have PSTN interfaces located in China, now the purpose is to FW calls
through PSTN.
Assuming I have a client who is calling from Japan to my main switch in U
- Non-Commercial Discussion
Subject: Re: [asterisk-users] Networking Concept
B.Masoud @ SH wrote:
> Assume I have a Main Asterisk Server located in UK, and another box that
> have PSTN interfaces located in China, now the purpose is to FW calls
> through PSTN.
>
> Assuming I have
- Original Message -
From: B.Masoud @ SH <mailto:i...@saudihome.com>
To: 'Asterisk Users Mailing List - Non-Commercial
<mailto:asterisk-users@lists.digium.com> Discussion'
Sent: Tuesday, October 06, 2009 1:14 AM
Subject: [asterisk-users] Networking Concept
H
, October 05, 2009 10:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] tdm outgoing
B.Masoud @ SH schrieb:
> I have defined the card g0 to have 24 channels, but
> every time I try to call, if all ports are off the call always go to the
> first
I have seen this message " stopped sounds " while I am watching asterisk
debug:
-- Called 9/0532828384
-- Call accepted by 192.168.10.220 (format ulaw)
-- Format for call is ulaw
-- IAX2/9-69 stopped sounds
-- IAX2/9-69 answered SIP/xxx.xxx.xxx.xxx-b7d009a0
What do
Hello all,
Assuming I have 1 asterisk with 24 channels fxo and another 2 asterisk boxes
all connected iax2,
I want to grand the first asterisk box to use all the 24 channels on the
main, but I want the 2nd asterisk to use only 8 port, how can limit the
second box from receiving more than 8
Anyone pls
I have seen this message " stopped sounds " while I am watching asterisk
debug:
-- Called 9/0532828384
-- Call accepted by 192.168.10.220 (format ulaw)
-- Format for call is ulaw
-- IAX2/9-69 stopped sounds
-- IAX2/9-69 answered SIP/xxx.xxx.xxx.xxx-
Sent: Thursday, October 08, 2009 12:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] limiting number of channels to be accessed
B.Masoud @ SH wrote:
> I want to grand the first asterisk box to use all the 24 channels on the
> main, but I want the
2:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] limiting number of channels to be accessed
/etc/asterisk/extensions.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Thursday,
Hello,
Sometimes the call gets answered for 1 second, but actually the phone has
not rang, its just the CDR, and asterisk hangup automatically, I cought the
log of 1 call like this, I hope you can help me with this.
My setup is :SIP--à ßIAX2---à
Here:
-- Executin
Hi
After a day of running asterisk, I got choppy sound when fw ip->pstn
When I restart asterisk the sound is fine,
Anyone had same problem?
Thanks.
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so I do a
"restart when convenient" each morning around 4:00 AM.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Friday, October 09, 2009 3:46 PM
To: 'Asterisk Users Mailing List - Non-Comme
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Friday, October 09, 2009 3:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] choppy sound
Hi
After a day of running asterisk, I got choppy sound wh
Hello all,
Is there anyway that I can configure Asterisk to start dialing out from fxo
after (xx) seconds from getting the dial tone? I don't want tdm card to send
the number immediately because it fails many times.
Thanks for any help.
___
-- B
On Behalf Of Doug Lytle
Sent: Saturday, October 10, 2009 5:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
>
> Hello all,
>
> Is there anyway that I can configure Asterisk to start dialing out
> from fxo
I have done the changes
exten => s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
I am getting this:
-- Executing [...@macro-dialout-trunk:19] Dial("IAX2/9-11592",
"DAHDI/r0/0559857826|300|") in new stack
-- Called r0/0559857826
Is it now on work? Or I have to restart?
Thanks.
-Origin
users] delay to dial
B.Masoud @ SH wrote:
> I have done the changes
> exten => s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
>
> I am getting this:
>
> -- Executing [...@macro-dialout-trunk:19] Dial("IAX2/9-11592",
> "DAHDI/r0/0559857826|300|") in new stac
Is there a ready add-on to asterisk that will display the ACD/ASR per
channel, source & destination?
Thanks.
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Register Now: htt
I am not sure why I am getting this message,
I have an outbound route that goes to asterisk gateway1 then asterisk
gateway2
When all lines on asterisk gateway1 are full, I get the message " all our
circuits are busy now" then few second later, the phone rings, going to the
second route! And the
I have tried more than 10 different branded/non branded, audiocodes was by
far the best fxo device..
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Wednesday, October 21, 2009 12:39 AM
To: Asteris
awn extension (macro-hangupcall, s, 11) exited non-zero on
'IAX2/9-16336' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on
'IAX2/9-16336' in macro 'outisbusy'
== Spawn extension (from-internal, 0505103250, 5) exited non-zero
Hello,
I have :
answeronpolarityswitch=yes
on chan_dahdi.conf
but it's making all my lines answer on polarity reversal, this causes a
problem for PSTN lines, so how can I set these lines to answer immediately
(when it rings)?
thanks
_
m.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese
Sent: Wednesday, October 21, 2009 10:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] polarity on some channels
B.Masoud @ SH wrote:
Hello,
I have :
answeronpolaritys
When Asterisk establish a call through an outbound trunk, Is there any way I
can know who hang up the call first? The caller or the party called?
Thanks.
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asterisk-users mailin
I have a trunk, and its host=dynamic dns.
The problem is, when the IP changes the
Sip show peers
Still show the old IP of the DNS, I have to reload and save the
configuration again so that asterisk recognize the new IP of the DNS.
Any idea how to automate such a thing? Or how can I keep as
Commercial Discussion
Subject: Re: [asterisk-users] Dynamic DNS trunk
If the trunk is a dynamic IP you need the other end to register to Asterisk,
so letting Asterisk know the new IP.
Regards,
Juan
B.Masoud @ SH wrote:
I have a trunk, and its host=dynamic dns.
The problem is, when the IP changes
digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Friday, October 30, 2009 1:53 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Dynamic DNS trunk
On 30/10/09 6:42 AM, B.Masoud @ SH wrote:
> Hi
>
> I tried with registration, it di
Hello
I just found out this:
I had a phone into the FXO ports to see why calls are not passing through,
When I ask asterisk to dial a number of 10 digits, it dials the first 9
digits, then wait 2 seconds and dial the last digit!
Any idea how to overcome this and dial the whole number 1 sho
Hello,
My client customers complaining that their calls suddenly get hung-up, I am
just investigating if the problem from my side, I had a log of a hang-up
case,
Does it help to know if there is a problem that can be resolved from my
side?
elastix*CLI>
-- Hungup 'IAX2/9-6813
In that case, you should be getting a message saying "hanging up call ,
no reply to our critical package". see if you receive a message like that in
your debugging.
CS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Ma
f the existence of a firewall
which can be blocking any necessary ports for tcp and/or udp.
You should always cover your real IP numbers when showing samples of your
logs
CS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.M
dnsmgr.conf:
enable=yes
refreshinterval=300
regards.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Friday, October 30, 2009 3:28 AM
To: 'Asterisk Users Mailing List
I have 2 questions:
1. Can I make outbound route rule based on the Source Channel?
2. Can I auto change the outbound route based on time/Day of week?
Any help very appreciated..
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Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] outbound routing
--
Sent from mobile device
On Nov 8, 2009, at 2:13 PM, "B.Masoud @ SH" wrote:
> I have 2 questions:
>
>
>
> 1. Can I make outbound route rule ba
Hello,
I would like to know how the following scenario works:
I have 3 Asterisk servers, A,B & C, each one is located in a different
country.
Asterisk A is the main one, and both B & C are connected to it.
My question is, when a call is originated from B to C, it will have to go
through
So how can I let A makes a PEER connection between B & C, and ONLY log the
call information?
Thanks.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Thursday, November 12, 2009 6:10 PM
To: Asterisk Users Mailing Lis
. If you don't have "NAT traversal Kung-Fu", I suggest using
IAX2 over SIP.
-K
- Original Message -----
From: B.Masoud @ SH <mailto:i...@saudihome.com>
To: 'Asterisk Users Mailing List - <mailto:asterisk-users@lists.digium.com>
Non-Commercial
Hello all,
How can I ask Asterisk to ignore a sip hang-up request for XX seconds from
the beginning of the session?
Thank you
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How can I set a maximum call duration on a ZAP channel?
Thank you.
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9 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] max call duration
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
B.Masoud @ SH wrote:
> How can I set a maximum call duration on a ZAP channel?
>
Look at the parameters on the Dial applicatio
Anyone can recommend a commercial large scale IVR with easy + pro management
for asterisk?
Thanks.
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Hello, I would appreciate if someone can give some help on what I want:
When someone call my box (from outside), to a certain ZAP port, it will put
him on hold, and immediately the box calls to outside SIP trunk to a
preconfigured certain number, then when the other party picks up the phone,
bo
Hello,
I would like to connect 2 asterisk boxes together, so this is my scenario:
Asterisk Main: it is connected to many sip providers and its main purpose as
a call termination forwarder.
Asterisk B: its connected to E1, and its purpose to terminate calls. It
will receive SIP messages
Was my question not understood?
Hello,
I would like to connect 2 asterisk boxes together, so this is my scenario:
Asterisk Main: it is connected to many sip providers and its main purpose as
a call termination forwarder.
Asterisk B: its connected to E1, and its purpose to ter
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