Hello,
I'm using Asterisk 1.6.2.10. Whenever I dial Local channels via asterisk
manager, the calls never get a hangup signal even with timeout specified. I
find channels with ZOMBIE text appended.
It ends up occupying all the channels with the result that asterisk thinks
every channel is busy,
thnks for the reply..
i want to know is there any way to call a SIP to SIP by command line
regards
Upendra
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On Thursday 01 March 2012, upendra wrote:
thnks for the reply..
i want to know is there any way to call a SIP to SIP by command line
Yes. Just write a script in your favourite language (even bash will do if
there is nothing better) to set up a callfile, then invoke it from the
I have a server with an OpenVox A400P card with 2 FXO modules on it. The
internal extensions are SIP Grandstream phones. When making or receiving
external calls through PSTN, there is an interrupted hissing like high
pitch noise - which might go away for few seconds then start again.
1. The
In almost all major releases of asterisk 1.6.x, SS7 Disposition never sets
to ANSWERED, even when someone answers the call, it logs NO ANSWER in
the cdrs.
Please help me resolve the issue.
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Hi team,
I am experience the same issue.
Thanks
Vinod dharashive
Sent from BlackBerry® on Airtel
-Original Message-
From: [Digital^Dude] ® millennium@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 1 Mar 2012 15:32:41
To: Asterisk Users Mailing List -
What versions on Asterisk and chan_ss7 are you using?
On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote:
Hi team,
I am experience the same issue.
Thanks
Vinod dharashive
Sent from BlackBerry® on Airtel
-Original Message-
From: [Digital^Dude] ®
Are you using AMI originate for these SS7 outbound calls?
On Thu, Mar 1, 2012 at 6:15 PM, [Digital^Dude] ®
millennium@gmail.comwrote:
What versions on Asterisk and chan_ss7 are you using?
On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote:
Hi team,
I am
Hi ,
Yes, I am using asterisk-java ami to originate call.
Using LibSS7
Thanks
Vinod dharashive
Sent from BlackBerry® on Airtel
-Original Message-
From: [Digital^Dude] ® millennium@gmail.com
Date: Thu, 1 Mar 2012 18:23:47
To: vdharash...@gmail.com; Asterisk Users Mailing List -
Howdy,
I have tried all of these and a few more. PBXinaFlash gave me the
best results, by far. AsteriskNow produced a basic working system. I
could not get any of the others configured to work at all. I should
tell you my restrictions. I was evaluating these distros to see which
one I could
I tried it on asterisk 1.8, and it worked fine.
On Thu, Mar 1, 2012 at 6:39 PM, Vinod Dharashive vdharash...@gmail.comwrote:
**
Hi ,
Yes, I am using asterisk-java ami to originate call.
Using LibSS7
Thanks
Vinod dharashive
Sent from BlackBerry® on Airtel
On 02/29/2012 02:28 PM, Dmitry Melekhov wrote:
btw, played with res_fax.conf
if I set maxrate=7200 fax machines try (and fail) 9600 anyway.
Why? If limited ti 7200? looks like bug...
Why do you think everything you don't understand is a bug? What you see
is correct behaviour. Any party in the
On Thu, 1 Mar 2012, Ralph Green wrote:
Howdy,
I have tried all of these and a few more. PBXinaFlash gave me the
best results, by far. AsteriskNow produced a basic working system. I
could not get any of the others configured to work at all. I should
tell you my restrictions. I was
Tom you're killing me with the me's please!
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On Thu, Mar 1, 2012 at 10:35 AM, Nick Khamis sym...@gmail.com wrote:
Tom you're killing me with the me's please!
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hahahaha!!
I've tried Elastix and FreePBX, for almost 4 years. Both are excellent!!
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Hi ,
I am using asterisk 1.6.1, any idea patch for the same
Thanks
Vinod dharashive
Sent from BlackBerry® on Airtel
-Original Message-
From: [Digital^Dude] ® millennium@gmail.com
Date: Thu, 1 Mar 2012 19:58:13
To: vdharash...@gmail.com
Cc:
On Thursday 01 March 2012, Ralph Green wrote:
Howdy,
I have tried all of these and a few more. PBXinaFlash gave me the
best results, by far. AsteriskNow produced a basic working system. I
could not get any of the others configured to work at all. I should
tell you my restrictions. I
Hello,
I am using a perl script to pull call info from a DB and place calls via
telnet and AMI, all on local machine of course. My problem is that I
need to capture any response from the carier, such as this taht appears
in the CLI:
[Mar 1 12:55:50] == Using SIP RTP CoS mark 5
[Mar 1
We are looking to find someone that is familiar with Fujitsu and Mitel
PBX's. Email ru...@inline.net off list.
Thanks
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A J Stiles wrote:
On Thursday 01 March 2012, Ralph Green wrote:
Howdy,
I have tried all of these and a few more. PBXinaFlash gave me the
best results, by far. AsteriskNow produced a basic working system. I
could not get any of the others configured to work at all. I should
tell you
5. Placing ferrite cores on the phone cables.
Do either of the phone lines in question have DSL on them?
If so, a ferrite core (which will block common-mode RF
signals) probably won't help much, if at all. DSL is a
differential-mode signal, and its frequency content starts
down in the tens of
I have been playing with gvoice over the past few months and it's been great
except for this error that appears ONLY when my firewall is enabled:
[Mar 1 14:08:19] WARNING[26490]: stun.c:406 ast_stun_request:
ast_stun_request send #0 failed error -1, retry
[Mar 1 14:08:19] WARNING[26490]:
Since you are using AMI, I would assume you are using one of the AMI
interfaces from CPAN or somewhere. If this is the case you could do
something like this:
my $astman = new Asterisk::Manager;
$astman-user('mickey');
$astman-secret('mouse');
my
All my phones have call waiting, so it's unlikely DIALSTATUS ever gets set to
BUSY. So, I'm trying to decide what to do about the two greetings users record,
busy and unavailable.
If I could, I could just disable one. Then there's only one greeting, and no
chance for confusion. I could
Howdy All,
I'm considering Asterisk / Digium as a replacement to my existing phone
switch. I need to continue to be able to push analog lines between
multiple buildings in a campus environment.
The Digium Analog 410 Card manual states it's not recommended to go
beyond 1500 feet distance for
On 2 March 2012 01:43, A J Stiles asterisk_l...@earthshod.co.uk wrote:
Look, the command line is a fact of life. Microsoft have spent a fortune
telling you that you're not smart enough to use it. You do not have to fall
for that. Are you going to sit back and let them call you stupid?
Hi all,
It disturbs me to see asterisk (v 1.6.2.10) writing CDRs even when there
are 0 active channels and 0 active calls. Is there an upper limit in terms
of CDRs / second that asterisk can handle? Does it queue the unwritten CDRs
somewhere?
Please help me clarify this confusion.
Thanks
--
On 01/03/12 19:07, Dave Platt wrote:
5. Placing ferrite cores on the phone cables.
Do either of the phone lines in question have DSL on them?
If so, a ferrite core (which will block common-mode RF
signals) probably won't help much, if at all. DSL is a
differential-mode signal, and its
We are running FreeTDM on a very cheap Atcom card, which used another
module ph_tor3_e1 on top of Dahdi. I believe this is derived from Torrenta.
http://www.atcom.cn/downloads/TelephonyCard/drivers/AX-4ET/E1/ph_tor3_e1.c
On Ubuntu 10.04 this gives problem as the module ph_tor3_e1 (and hence
Asterisk can cache cdr records to avoid having to write continuosly in the
cdr backend. Writing in bunch instead one at once improves performance.
Check the cdr.conf file and disable the option batch if it hurts you.
Leandro
Il giorno 02/mar/2012 07:24, [Digital^Dude] ® millennium@gmail.com
I've tried with batch enabled as well as disabled, it seems irrespective of
the call burst I send to asterisk. CDR writes at a constant speed, not
changing with the call load!
On Fri, Mar 2, 2012 at 12:20 PM, Leandro Dardini ldard...@gmail.com wrote:
Asterisk can cache cdr records to avoid
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