Displaying 20 results from an estimated 3000 matches similar to: "how asterisk knows which context forward the call to?"
2009 Nov 07
4
Help with concurrent VoIP calls
Hi. I'm having trouble figuring out why I'm not able to make many
concurrent VoIP calls on my system. I'm not aiming for a huge number,
because I have purposely bought a low powered system, but I would
think that I could get more. Here are the details:
I have a small-form-factor Asterisk server with an Intel Atom 230 CPU
(1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu
2010 Jul 01
2
Brute force attacks
Hi
We've just noticed attempts (close to 200000 attempts, sequential peer
numbers) at guessing peers on 2 of out servers and thought I'd share the
originating IPs with the list in case anyone wants to firewall them as
we have done
109.170.106.59
112.142.55.18
124.157.161.67
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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2013 May 12
3
time zone setting in asterisk
Which file in Asterisk have a setting for time zone?
When asterisk record incoming call in Master.csv the time is 6hr. ahead.
I'm on: Canada/Mountain zone
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Joseph
2014 Dec 23
4
Connect Asterisk to WiFi
Are there any adapters that would allow me to connect asterisk to wifi or we are not there yet?
I have Digium adapter S101i that was discontinued but similar device that would connect to wifi network and a cell phone would be handy.
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Joseph
2008 Aug 20
2
Linksys SPA3102-NA firmware upgrade on Linux
Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org
http://www.voip-info.org/wiki/view/Sipura
--
#Joseph
GPG KeyID: ED0E1FB7
2012 Jan 07
2
Asterisk 10.0 & 1.4 - iax codec are not compatible
I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better.
I'm still getting:
WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have <11>, digest has <pstn-1270>
NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c1222950155
Anybody
2014 Dec 24
2
Connect Asterisk to WiFi
On Tue, Dec 23, 2014 at 6:34 PM, Rusty Newton <rnewton at digium.com> wrote:
> On Tue, Dec 23, 2014 at 4:17 PM, Joseph <syscon780 at gmail.com> wrote:
>> Are there any adapters that would allow me to connect asterisk to wifi or we
>> are not there yet?
>> I have Digium adapter S101i that was discontinued but similar device that
>> would connect to wifi
2009 Dec 11
3
ATA FXO
I'm looking for a reliable ATA FXO/FXS adapter.
Linksys 3102 - a lot of echo problem + two of them died within a year (not reliable)
Sangoma USBFXO - problem installing drive in Gentoo.
I've tried two Chines units: AG-188N and YGW30B
none are of them have real FXO port that will register with Asterisk.
Any other recommendations; (I don't like internal cards).
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Joseph
2012 Nov 22
3
monitoring asteriks
How can I monitor asterisk if all lines are registered etc?
I have an asterisk on a remote location and sometime they reporting problems that phone is not ringing, they can not dial out etc.
Usually I just restart asterisk and it solves the problem.
Is there an application that will email me if case any line looses registration with with asterisk?
Or any better solution!
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Joseph
2010 Apr 01
0
Question about MaxRetries in the Asterisk Outgoing folder
I'm doing some automated calling by putting .call files in the Outgoing
folder of Asterisk. I'm concerned this might be a stupid question, but I'm
pretty sure I've done my research well and I'm unable to come up with an
answer on my own.
I want to know: what happens to the .call files after the "MaxRetries"
number has been reached?
In my experience, they stay in the
2014 Dec 24
1
Connect Asterisk to WiFi
On Tue, Dec 23, 2014 at 6:51 PM, Joseph <syscon780 at gmail.com> wrote:
>
>
> Most cell phone don't have a USB port but you are correct, maybe I just need
> IAX2 soft-phone like:
> Zoiper - it works on most of the platforms. I think Zoiper registers
> directly with Asterisk IAX2 (if configured) as an extension, isn't it?
If your cellphone is capable of a Wi-Fi
2008 Aug 17
2
Running asterisk as non root user
Hi,
I've followed instructions of the book "AsteriskFutureOf TelephonySecEdit" on page 295 onwards ) Link to the Asterisk book: http://downloads.oreilly.com/books/9780596510480.pdf) and get an error when running service asterisk start. The error is: cat: /var/run/asterisk.pid: No such file or directory . I can run aserisk fine from the non-root user. Please help
Code Snippet:
1:
2011 Sep 03
1
upgrading from 1.4.39 to 1.8.5
What sort of things should I watch out for when upgrading from 1.4.39 to 1.8.5
Thanks,
--
Joseph
2010 Dec 29
2
GotoIf CALLERID(num)
I'm testing GotoIf($["${CALLERID(num) but I'm missing something as it is not working:
[office-open]
exten => s,1,Wait(1)
exten => s,2,Answer()
; for Caller ID is 471-5665, always signal congestion:
exten => s,3,GotoIf($["${CALLERID(num)}" = "4715665"]?4:6)
exten => s,4,Playtones(congestion)
exten => s,5,Congestion(5)
exten =>
2009 Oct 19
1
Cisco 1751 setup with asterisk
How hard is to setup Cisco 1751 w/2x FXO with asterisk?
I was googling but couldn't find much information; how to access unit interface for programming?
It might be a good replacement for Linksys.
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Joseph
2013 Jun 20
1
asterisk -rx "core show channels" + time
When I type: asterisk -rx "core show channels"
I usually get
Channel Location State Application(Data)
SIP/pstn-4444-000003 7807574622 at internal: Up Dial(SIP/77807574622 at pstn-9998
SIP/pstn-9998-000003 (None) Up AppDial((Outgoing Line))
Is there a way to pull information about time the channel started?
--
Joseph
2014 Sep 18
1
conversation record prematurely
I have following line in a context:
...
exten => _587NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten => _587NXXXXXX,n,MixMonitor(${recordfilename},b)
...
It records the conversation but it ends prematurely, after 10min. Why?
Where is the setting to records until a user hangup the handset.
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Joseph
2014 Sep 11
3
if statement recording - after hours
In my dial plan I have these two lines:
exten => _NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav)
exten => _NXXXXXX,n,MixMonitor(${recordfilename},b)
How to add "if" statement to execute these line only after let say 5pm. To record conversation only after 5pm.
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Joseph
2008 Nov 15
3
IAX2 client for "eee pc 1000"
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)?
I'll eventually replace this crippled Linux with something better but I don't time to play around with it as most divers and modules are still too new and
not fully available in all distros.
--
#Joseph
GPG KeyID: ED0E1FB7
2008 Sep 02
4
AgentCallbackLogin AddQueueMember
Hi
i have problem with AddQueueMember logic.
I need login Agent(Member) in asterisk.
use this option:
for example:
AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13)
and now i want to call to this Agent:
exten => _1XX,1,Dial(Agent/${EXTEN:1})
call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga.
This doesn't work, How can i do this on Asterisk 1.4(not