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
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
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,
Anyone can recommend a commercial large scale IVR with easy + pro management
for asterisk?
Thanks.
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To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] max call duration
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B.Masoud @ SH wrote:
How can I set a maximum call duration on a ZAP channel?
Look at the parameters on the Dial application.
Barry
How can I set a maximum call duration on a ZAP channel?
Thank you.
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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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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
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 Discussion'
Sent: Thursday, November 12, 2009 6:10
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 i...@saudihome.com wrote:
I have 2 questions:
1. Can I make outbound route rule based on the Source
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 - Non
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
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Saturday, October 31, 2009 8:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Calls disconnects after short time
Hello,
My
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Sunday, November 01, 2009 12:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Calls disconnects after short time
My server use public ip, so
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
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 the
Sip show
[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 did not update the IP address
I
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
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
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
[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 :
answeronpolarityswitch
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:
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Tuesday, October 20, 2009 4:35 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] all our circuits are busy now
On 20/10/09 1:30 PM, B.Masoud @ SH
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
Is there a ready add-on to asterisk that will display the ACD/ASR per
channel, source destination?
Thanks.
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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.
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] 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 after (xx) seconds from
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.
-Original
.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
Sent: Saturday, October 10, 2009 9:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] delay to dial
B.Masoud
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 : vendor SIP--à Asterisk ßIAX2---à Asterisk with
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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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-Commercial Discussion'
Subject
[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 when fw ip-pstn
When I restart
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
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
, 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^nd asterisk to use only 8
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, October 08
, 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 port, how can I
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 does it
- 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 a client who is calling
- 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
Hello,
I would like
send a call to the lines attached
to the card?
PaulH
B.Masoud @ SH wrote:
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
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
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
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
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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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???
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-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
Alba Fotónica S.L
Is there any program Asterisk users use to calculate ASR and ACD ??
Thanks for any comments.
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[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 and ACD ??
Thanks for any comments
[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 appreciated, I know about
Can you provide me some code for that?
I am NOOB
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Wednesday, September 09, 2009 5:31 PM
To: Asterisk Users Mailing List - Non-Commercial
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:
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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] On 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 signal,
reset CDR
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