Displaying 20 results from an estimated 3000 matches similar to: "Re: Generate Random Numbers in dialplan"
2008 Aug 13
1
Sending Set Asynchronous Balanced Mode Extended
So we've got a TE410P configured as E-1. The PRI is showing up as normal, I have green lights, but d channel doesnt seem to come up and i keep getting this error if i do a "pri intense debug"
The carrier swears up and down that there are no issues on their end. Any thoughts?
localhost*CLI>
> Unnumbered frame:
> SAPI: 00 C/R: 0 EA: 0
> TEI: 000 EA: 1
> M3:
2006 Oct 14
1
Re: Generate Random Numbers in dialplan
On Sat, 2006-10-14 at 12:00 -0700,
asterisk-users-request@lists.digium.com wrote:
> Steve,
>
> Is RAND available in the latest trunk or do I need the 1.4
> beta?
>
> If I do show function RAND it says its not available.
>
> Thanks,
> Jon
Jon--
Forgive me, you didn't say which version you
2006 Jan 31
2
Canadian Termination $0.0039 / Minute
All we have a deal on Canadian termination.
Rate: $0.0039 US Dollars
Billing: 1/1
Protocol: SIP or H323
Codec: G729
Terms: Prepaid Only.
We have a real-time web interface where you can monitor or download your CDR's.
Please e-mail me offlist if you are interested: jweisman@ibell.net
Thanks,
Jon
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2019 Mar 25
3
[Bug 1328] New: Please allow ipset add and del via the /proc/net/xt_ipset mechanism
https://bugzilla.netfilter.org/show_bug.cgi?id=1328
Bug ID: 1328
Summary: Please allow ipset add and del via the
/proc/net/xt_ipset mechanism
Product: ipset
Version: unspecified
Hardware: x86_64
OS: All
Status: NEW
Severity: enhancement
Priority: P5
Component:
2010 Nov 22
3
Is existing CDR in Asterisk is enough for complete billing
Hi everyone,
I am facing lots for problem with CDRs in 1.6 and above
versions,its shows wrong records when I do transfer(caller side and
calee side),callforward,call parking.Is the present CDRs in 1.6 is
enough for Complete billing.?What I need to do to make it proper.Please
help me on this.
Thanks
Nikhil
2010 Nov 07
3
Why are the hackers scanning for these?
Hey, I'm going thru logs, and I see some very common and interesting things
that the hackers are looking for.
In a whole bunch of scans, I've noticed that the first guess or two for sip
accounts
is usually a 10-digit number. I'm asking myself, why these numbers? Are they
looking
for a voip trunk? Or is it just like a serial number for the scan? What?
Here's some examples:
2016 Sep 08
3
PJSIP Weirdness, or just my weirdness?
Hello!
Oh, wise ones, ponder with me over two of the surprises that
populate the universe!
I have a phone, that I sometimes cannot reach, connected via pjsip.
It can call other extensions just fine, it can call out over a
trunk to my cell, all is well, but getting a call? Forget it most of the
time.
Here is all the config relevant to that phone:
[murftest12]
type=aor
qualify_frequency=1992
2019 Jul 05
2
unsolved: Re: solved: how to create a working certificate for using TLS?
On 7/5/19 10:50 AM, Doug Lytle wrote:
> On 7/4/19 6:40 PM, hw wrote:
>> This has again, and for no reason, ceased to work again after
>> restarting asterisk. No matter what I try, I can't create a
>> certificate asterisk
>> would verify.
>
> Have you considered using LetsEncrypt for a valid certificate?
>
> Doug
>
>
What would be the point
2007 Jun 26
0
Slip Events
All,
I'm using a Digium TE410P w/ Asterisk 1.2.18. Trying to connect it to our NACT STX switch via PRI, d channel is up, T1 shows normal, but I'm getting crazy errors. I rewired this thing three times, then I connected the same cable from the STX to a Cisco AS5300 (same pri settings as asterisk), and all slip events and frame sync errors went away, so the cable is good.
STX --> DSX
2010 Dec 11
1
No more room in scheduler
Dears:
Really, later I faced problem in the asterisk system which is :
Message is shown when the unique id which is generated with each caller reach
9000 and something:
No more room in scheduler
Asked to delete sched id
.
.
after I restarted the server this message is not shown again till now (after 2 week)
>>>
My question:
What is the reason of this error and how can I solve the
2007 Aug 02
1
asterisk1.2 to 1.4 g711a fax
hi,
i have problem with pass-through faxing
with this scenario
hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen
virtual) - linksys ATA
i can fax to fax2mail on hylafax
but after upgrade asterisk2 to 1.4 faxing is not working
hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen
virtual) - linksys ATA
configuration is same
do you hava any idea what is
2007 May 23
1
CDR on channel 'IAX2/u92613106-3' already started
Hi all,
I'm having a problem with an asterisk server being unable to call certain
cellphones and answering machines. Anytime the person answers the phone
call, everything works well. But when the call goes to voicemail or an
answering machine, I get the error message below:
=====================================
*CLI> -- Attempting call on IAX2/u92613106/15551234567 for
2009 Jun 04
1
CDR question
Hi,
Asterisk does not post CDR when dial status is CHANUNAVAIL.
Can someone tell me what are the conditions under which CDR is not posted?
Thanks
Jim
2015 Mar 19
1
Asterisk 13 : SILK codec ?
On Wed, Oct 29, 2014 at 7:10 PM, sean darcy <seandarcy2 at gmail.com> wrote:
> On 10/29/2014 08:06 PM, Matthew Jordan wrote:
>
>> On Wed, Oct 29, 2014 at 5:16 PM, sean darcy <seandarcy2 at gmail.com> wrote:
>>
>>> Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13?
>>>
>>>
>> codec_silk for Asterisk 12 will most
2007 Oct 16
7
E4 Superframe E&M?
I need to create a couple of tie lines between a legacy system and an
Asterisk system. I was told that the tie lines are E4 Superframe E&M.
I have done E&M wink but have no idea about E4 Superframe E&M and Google
is not helping me here.
Does anyone know about this type of signaling and if Asterisk can handle it?
Thanks,
Steve
2008 Sep 26
0
PRI TE110P Configuration (Solved)
Hi,
The problem solved
After installing new zaptel drivers, we ran the "genzaptel" command to
generate /etc/zaptel.conf file,checked with "zttool" command and the card
status was "Yellow alarm/Blue alarm/Recovering" and the card LED was
blinking red and green.
The problem was with the generated zaptel configuration., but not with
the pin
2007 Apr 16
6
BSNL caller ID (India)
Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2
I have tried following link
http://bugs.digium.com/view.php?id=6683&nbn=24
but was not able to get it, although did not ge any error too.
I always get the caller id as asterisk.
Can someone please help.
Regards,
Sanjay Rajdev
2011 Dec 27
1
how to used SIPp for sip load testing
Hi list,
I have installed SIPp into my server. But not able to used it properly.
how to configure with my server ? how to see logs on webpage ?
how to start call testing ....
when i start SIPp then found verious hits on myserver.
*CLI:- *
[Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite:
Call from '' to extension 'service' rejected because extension not
2010 Jan 17
1
receive text
Is there any code that I can cut/paste that will allow me to receive
an SMS text on Asterisk?
and, where can I capture the incoming text?
2006 Mar 19
7
An FXO version of IAXy?
Hello--
In the interest of Symmetry, does anyone else in the world see any need
for a device like the IAXy (or the SIP ones from other manufacturers,
like the ATA186), but one that presents an FXO interface instead, so it
can be connected not to phones, but the PSTN?
murf