On 20 Aug 2015, at 11:12, Sevana Oy sa...@sevana.fi wrote:
Curious why didn’t you try AQuA http://sevana.biz/products/aqua/ to score
the quality? Using voice files for tests has more representation to my
opinion.
Spot the salesman? ;)
Steve--
On 11 Mar 2015, at 17:53, Matthew Jordan mjor...@digium.com wrote:
On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes
steve-li...@geekinter.net wrote:
Anyone know where it’s gone?.. Appears to have been down all day.
The hamsters should be running in their wheels again now.
Cheers Matthew. Give
Anyone know where it’s gone?.. Appears to have been down all day.
Steve
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On 9 Feb 2015, at 15:32, Francisco Leonardo Mota francisco.m...@rnp.br wrote:
Submission.
Thanks,
Uh, no problem?..
Steve
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On 10 Nov 2014, at 13:01, Norman Laidla norman.lai...@telegrupp.ee wrote:
Well, pants. It actually is causing a problem, because the phone doesn't use
any other methods to register to Asterisk. This is a bit of a big issue.
What softphone is it? It sounds like rather odd behaviour.
--
The Asterisk 13 is already stable for production environment?
It’s only been out a couple of days - hard to make judgements just yet. But it
is out of Beta so should be of reasonable quality - give it a try :)
Steve
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_
--
On 7 Oct 2014, at 09:24, Dania Asi da...@futuretrendsest.com wrote:
Kindly note that I asked about the capability of the phones and now I am
asking about the way I can do it to my client's phones, because he is asking
for a demonstration.
Yet you’ve not even told us the phones in use. You
On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be wrote:
So just before hanging up, I add a custom SIP-header :
exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan)
exten = s,n,Hangup()
SIPAddHeader only works for INVITE as far as I know.
Steve--
On 2 Sep 2014, at 10:38, Jonas Kellens jonas.kell...@telenet.be wrote:
Then how can I let another Asterisk server know the custom reason of hangup ?
If it is not possible with custom SIP-header, then how ?
As far as I know that’s going to require a source change. May not be the case
with
On 18 Aug 2014, at 09:27, Усин Айбек prince...@gmail.com wrote:
I have trouble with connection to AMI 1.1 wich enabled on Elastix
Asterisk Call Manager/1.1
Action: Login Username: admin Secret: qweasd123
Response: Error
Message: Missing action in request
You are missing the newline
On 8 Aug 2014, at 06:05, Gergo Csibra csi...@gmail.com wrote:
back in the old analog telephony days there was digital PBX-es and
digital system phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension
On 5 Aug 2014, at 17:10, Mike Diehl mdiehlena...@gmail.com wrote:
All of my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10.
If you do firmware upgrade your 8000s, don’t go past 6.1.3 or it’ll go badly…
Freezing and requiring power-cycle, clocks stopping (and showing minus
On 27 Jun 2014, at 15:37, Anurag Rana anuragrana31...@gmail.com wrote:
There are lot of requests coming in and I am not able to stop it because I am
unable to detect the IP address.
I used wireshark to capture the packets.
If you can capture the packet, surely you have the IP? If they intend
On 2 May 2014, at 10:07, upendra uppi...@gmail.com wrote:
Am new to Elastix and wanted to try build new modules in the Elastix , so i
want to know how the PHP is running ?? as i see no Apache server inside ?? so
wanted to know how its running ? which server and architecture?
This is not an
On 26 Mar 2014, at 15:05, Michelle Dupuis mdup...@ocg.ca wrote:
I see a lot of attempts by hackers to call 00972595301123 or
011972595115207 or variations but that same 972595 is often present.
Can someone break down that dial string with an explanation? The 011 look
like an overseas
On 26 Mar 2014, at 16:20, Michelle Dupuis mdup...@ocg.ca wrote:
If this is to 972 area code then the next digits should be 0X or 0XX but they
are not. This differs from what I found documented for that area code - I
thought someone from the region might add to the discussion. Not sure if
On 25 Mar 2014, at 14:16, Digium's Asterisk Development Team
asteriskt...@digium.com wrote:
We apparently have a spam bot subscribed to the list and replying
*directly* to anyone who posts on the list.
There’s plenty of people harvesting the list archives too, I get loads of spam
about
On 25 Mar 2014, at 15:00, Jeff LaCoursiere j...@jeff.net wrote:
On 03/24/2014 05:50 PM, Thorolf Godawa wrote:
But your carrier has to support T38, when we began to evaluate this some
years ago, this was not true for all.
Would you share the provider you are using? I have had almost zero
Probably should post this to the asterisk-biz list. This is the non-commercial
discussion list. Post to the -biz list and you’ll probably have loads of sales
droids happy to help :)
Steve
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On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote:
I recently experienced an odd situation. I have an Asterisk 11.5.0 system
(Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At
some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box
On 17 Jan 2014, at 02:18, Sean Darcy seandar...@gmail.com wrote:
I'm used to seeing fraudulent attempts to authenticate, But now I'm getting
them from the server itself.
I have an asterisk server behind a firewalled router. The local subnet is
10.10.10.0/24, the server is 10.10.10.100.
On 14 Jan 2014, at 02:19, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:
Thanks for your feedback Paul. The not having outbound trunks is going to be
a challenge.
Why? it’s what contexts were invented for.
Steve
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On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load
balance incoming calls over IAX2 trunks. If any trunk goes down the calls
traffic will be shared with other available trunks. When it gets Up the
On 18 Oct 2013, at 04:06, John T. Bittner j...@xaccel.net wrote:
Today I was hacked but caught it very quickly. This is the weird part, they
hacked an IP Auth based account by simply knowing the account name.
How is this possible? I am running Asterisk 11.5.0. Now it’s my fault I used
a
On 13 Sep 2013, at 11:44, A J Stiles wrote:
In the Windows world, where you usually don't get the Source Code, you never
know what is running on your computer; in which case, you are never sure that
there isn't a daemon listening on a particular port number, so it is wise in
that case not
On 27 Aug 2013, at 15:34, Ben Klang wrote:
But what's REALLY useful is Sippy Cup's ability to dynamically generate PCAP
audio. If you've ever needed to drive an IVR from SIPp you're probably
familiar with the pains - it usually requires capturing an actual call,
isolating the RTP, and
On 20 Aug 2013, at 12:25, Pat Collins wrote:
Here ya go:
112233# use ${EXTEN:0:6})
123# use ${EXTEN:0:3})
123456789# use ${EXTEN:0:9})
I think 'variable length' implied 'unknown length'...
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On 6 Aug 2013, at 19:28, Mike Diehl wrote:
We got it fixed! Our co-lo is in the process of doing a network
reconfiguration/relocation and had changed their MTU to 1400 during
the transition. Once we did the same, everything started to work.
PMTU should take care of that. Are you blocking
On 16 Jul 2013, at 04:10, Klaverstyn, David C wrote:
I’m hoping someone can recommend a method to integrate Microsoft CRM with
Asterisk. Preferably an open source product otherwise a commercial product.
Hi,
You've not said what you're trying to integrate... Creating tasks for calls,
contact
On 3 Jul 2013, at 12:28, I.Pavlov wrote:
[2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter:
Call to peer '0014' rejected due to usage limit of 1
-- Couldn't call 0014
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [0014@sub_pbxdialco:50]
On 23 May 2013, at 10:49, bilal ghayyad wrote:
Facebook and Whatsapp sort-of support XMPP, so we can use Jabber to
communicate with them. But, how much jabber channel in asterisk is stable and
updated?
You can find out the support status from menuselect
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
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On 26 Feb 2013, at 16:52, Gary Carr wrote:
Is it possible to issue the POKE to a end point from the CLI? Our asterisk
servers is not seeing some end points drop off and I would like to create a
script to manually check end points.
http://www.geekinter.net/iaxping.txt
May be of use to you.
On 18 Feb 2013, at 17:03, Christopher Harrington wrote:
On Mon, Feb 18, 2013 at 10:54 AM, Steve Edwards asterisk@sedwards.com
wrote:
) If the AGI is [...] a compiled executable (C, Fortran, Cobol, assembler,
etc.)
I'd like to see an AGI written using Fortran or Cobol.
Don't tempt me
On 28 Jan 2013, at 13:55, Steven Howes wrote:
Who do I need to poke to get the yum repository / RPM files updated? The
dahdi RPMs are not up to date with the CentOS kernel versions any more, it's
making doing an installation a bit tricky due to dependancies, I'd rather not
roll back
On 6 Feb 2013, at 20:06, Rusty Newton wrote:
- Original Message -
From: Steven Howes steve-li...@geekinter.net
On 28 Jan 2013, at 13:55, Steven Howes wrote:
Who do I need to poke to get the yum repository / RPM files
updated? The dahdi RPMs are not up to date with the CentOS kernel
On 4 Feb 2013, at 12:53, Jonas Kellens wrote:
I call with my cellphone to our public telephone number
Our receptionist answers the incoming call and does an attended transfer to
my colleague ( A )
Colleague answers and the receptionist tells him that I am on the other side.
Receptionist
On 4 Feb 2013, at 13:45, Jonas Kellens wrote:
The IP-phones in this case are Yealink T32G.
What setting is needed in this IP-phone ?
Quick google doesn't turn up any results. Handsets probably dont support it.
Steve--
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--
Hi All,
Who do I need to poke to get the yum repository / RPM files updated? The dahdi
RPMs are not up to date with the CentOS kernel versions any more, it's making
doing an installation a bit tricky due to dependancies, I'd rather not roll
back / remove new kernels if I don't have to..
On 10 Jan 2013, at 02:10, Jai Rangi wrote:
I have removed yours right away.
Yes, I agree, But just like any company we have purchased/collected email
from different source. Also just like any company we are not perfect, we make
mistakes.
Then buy your addresses from different sources,
On 3 Jan 2013, at 15:13, Michael L. Young wrote:
So, I am asking the community for any input. I have read on here and seen on
IRC that some in the community are successfully using Asterisk with Verizon
SIP. Verizon was going to check and see if they have any notes about that
and those
On 2 Jan 2013, at 15:54, Eric Wieling wrote:
On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen benny+use...@amorsen.dk
wrote:
There is a sanction. People like me will score top posters lower and
soon not see their posts at all.
I'm the opposite. I'm likely not to scroll down 10 pages to
We should all top *AND* bottom post!
On 31 Dec 2012, at 06:03, isr...@gmail.com wrote:
Just my pitch in to post
From a blackberry you can only top post there is no way of bottom posting
So if I would have to wait to get to a computer to bottom post I would just
never answer
We should all
On 10 Dec 2012, at 16:13, Christopher Harrington wrote:
On Mon, Dec 10, 2012 at 5:23 AM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Password= c3podb@2012
In case you didn't realize you were sending this out publicly to a publicly
archived and searchable list, you
Hi,
SIP registrations typically try to register, are them prompted for a password
(via a 401 message) it then sends a new request with authentication . This is
normal.
Steve
On 23 Oct 2012, at 13:26, Jerry Geis wrote:
I have a connection between two asterisk boxes, both running 1.4.43
On 22 Oct 2012, at 15:21, Ishfaq Malik wrote:
On Mon, 2012-10-22 at 19:17 +0500, Control Oye wrote:
I have successfully setup Asterisk realtime. Now I can create
extensions dynamically. But when I put this command on cli mode
sip show peers
it returns no result.
can any one guide me to
On 18 Oct 2012, at 16:50, Mitul Limbani wrote:
U would have to write a dahdi module for this 3G modem to help asterisk
understand it as standard gsm channel.
Look up chan_datacard (i think that's what it's called from memory).
Steve
--
On 20 Aug 2012, at 01:37, Noam Birnbaum wrote:
A client wants to keep their old Inter-Tel KTS analog phones for budget
reasons. Two questions:
1. How could they use these with FreePBX?
You'd need an ATA or an analogue card. And how well they work, depends on if
they are 'true' analogue, or
On 1 Aug 2012, at 07:05, Support wrote:
I've looked on Cisco's website and Googled around, but I can not find a true
example of a provisioning file for this device. Anything I could find would
be
enough for me to make a template.
Download the SPC tool from the Cisco site. It includes the
On 1 Aug 2012, at 12:39, D Tucny wrote:
HP refuse to let you put your data at risk, refusing to activate write
caching without a charged battery attached or NV cache.
I personally would like to have the option to override things like this at my
own risk, but, HP don't give you that
On 25 Jun 2012, at 16:58, Jeff LaCoursiere wrote:
Actually we get that complaint a lot too (Polycom ring volume). We
typically install in hotel environments, and in their back office the
environment can be noisy, as well as in their restaurants.
I imagine in a typical office environment
Thought this deserved a name and shame!
;)
Steve
Begin forwarded message:
From: Pavel Ismailov pavel.ismai...@gmail.com
Date: 27 April 2012 06:58:07 GMT+01:00
To: steve-li...@geekinter.net
Subject: Flashphoner
Hello!
My name is Pavel Ismailov
and I`m CEO of www.flashphoner.com
Can't tell if this is a transparent attempt at advertising, or...?
S
On 19 Apr 2012, at 22:09, Josué Conti wrote:
This is your website:
http://www.convergia.com/
Thanks in advanced for any informations.
Best Regards
Josue
Em 19 de abril de 2012 17:11, Josué Conti
On 3 Apr 2012, at 16:42, Vik Killa wrote:
#disasterisk fail
#freeswitch win
#unhelpful comment
S
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On 30 Mar 2012, at 10:14, Syco wrote:
Finally the problem is: I cannot manage more than 80 concurrent calls.
What happens on the 81st call?..
S
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On 30 Mar 2012, at 10:04, Sean McMaster wrote:
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
It's not tricky.. Really.. It's on the bottom of every email.--
On 2 Mar 2012, at 11:35, Kamlesh Kumar wrote:
$agi = new AGI();
$agi- exec('Background','press_one0press_two0press_zero0');
$agi- exec('Read','NUMBER,,1,3');
$agi- verbose (You have entered.$NUMBER);
You need to use AGI to read the Asterisk variable.. Asterisk variables don't
magically
On 13 Feb 2012, at 12:06, virendra bhati wrote:
You can't set callerid for outgoing calls in case of PRI.
Why not? Every PRI I have used supported it. Is this a carrier-specific thing?
S
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On 11 Feb 2012, at 13:41, Kevin P. Fleming wrote:
At this time, there's no way to do it directly in the dialplan
In extensions.conf
[globals]
SEMICOLON=;
Then use ${SEMICOLON}
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On 9 Feb 2012, at 11:08, Gilles wrote:
Does someone of a good site/blog that keeps track of new releases of
Asterisk, and explains what the major changes/features when they do
occur?
Why not just use the latest version?..
S
--
On 7 Feb 2012, at 14:27, Gilles wrote:
On Mon, 06 Feb 2012 10:24:42 -0600 (CST), Richard Mudgett
rmudg...@digium.com wrote:
The UPGRADE.txt and CHANGES files do just that. They have been a part
of the Asterisk source files for a long time.
Thanks for the info. The problem is that the
On 27 Jul 2011, at 17:11, CDR wrote:
This is turning into a political issue such as the one in Washington
and the impending default on US debt.
No, YOU are turning this into a political discussion.
The point is that a minor change
in the code would have a dramatic effect on security, and
Hi All,
Asterisk 1.6.2.19 was released on the 28th, does anyone know if there a
timescale for this reaching the RPM repository? We're badly affected by a bug
in previous versions that has only recently become apparent to us. It's in a
situation where rebuilding from source isn't too practical
On 20 Jun 2011, at 16:33, Sagbo Romaric wrote:
Any client behind his NAT can talk with another behind his NAT.
Still not possible.. The internet doesn't really work like that. SIP even more
so.
S--
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On 15 Jun 2011, at 11:20, mahesh katta wrote:
i DID SOME VOICE MAIL CONFIGURATION. SO HOW CAN YOU RETRIVE THAT VOICEMAIL.
WHEN I RETRIVE THE VOCIE MAIL ITS NOT GETTING ANY MSG
A lot of filesystems are case sensitive. Maybe you wrote your configuration in
caps? This would also explain why you
On 3 Jun 2011, at 11:57, devr devr wrote:
My query now is willl all voxbone numbers show up as the operator as Voxbone
SA as above. I wanted to find out who the service provider is on some
numbers, I suspected the service to be voxbone but the operator shows as
other companies.
My idea
On 1 Jun 2011, at 22:50, Jesse Thompson wrote:
We are managing an Asterisk installation for residential VOIP service, and we
are having a problem where all inbound calls to DIDs which are assigned to us
by our wholesaler but not yet assigned to a downstream customer get caught in
a routing
On 4 May 2011, at 15:01, vip killa wrote:
screw that i just got hylafax to work with IAXMODEM...i refuse to pay
digium a dime... supposed to be open-source right?
There is so much wrong with that sentence, I don't know where to start.
On 4 May 2011, at 16:02, vip killa wrote:
Honestly
On 21 Apr 2011, at 13:46, A J Stiles wrote:
You *might* be able to recover the document, *if and only if* the recording
quality is high enough. Easiest way to try it is to call up a fax machine
(either an actual real one, or a copy of Hylafax) from Asterisk and play the
wav file down
On 18 Apr 2011, at 11:06, bilal ghayyad wrote:
I am using Asterisk for Call Center (so agents login, logout, ready, not
ready, ... etc). To be able to have a good call center reporting, on what I
have to depend? On the CDR of Asterisk or there is another way?
Is there a good open source
On 11 Apr 2011, at 10:03, darin iv wrote:
please send me the ways to change asterisk port from 5060 to 5061 i
need to configure it because we are already using 5060 port in router
then we cant use it again we have to configure other sip server so
please suggest me a
On 11 Apr 2011, at 15:28, vip killa wrote:
I'm using voicemail ODBC with Asterisk 1.6.2.17.2.
Why do I see Length is 186545 or something similar but a different number
in Asterisk CLI everytime someone leaves a message?
Because not all messages are the same length. I'd guess it's length in
On 7 Apr 2011, at 14:32, Deka, Rajib IN MAA SL wrote:
Is the following is the link for getting the source,
http://svn.asterisk.org/svn/asterisk/trunk/
Please try not to reply to the entire digest..
S
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On 6 Apr 2011, at 11:54, Silver Thorne wrote:
Does anyone know of any opensource or otherwise solutions out there that I
can try out?
Asterisk. Google it. If you're too lazy, Google MixMonitor. If you're too lazy
for that:
http://www.voip-info.org/wiki/view/MixMonitor
S
--
On 6 Apr 2011, at 17:46, vip killa wrote:
I have externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl so that when
someone is left a voicemail it will call the person's mobile phone and prompt
them with the new message. The perl script simply originates a call to a
persons mobile phone and
On 31 Mar 2011, at 13:52, Sebastian wrote:
http://www.downforeveryoneorjustme.com/downloads.asterisk.org
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On 28 Mar 2011, at 14:19, vip killa wrote:
Yes I followed directions on that page
Running Asterisk 1.6.1.22, anybody else experiencing this?
How often does fail2ban check the logs? It can only block that often, so if
more attempts happen in that time period it can't do anything until it knows.
On 24 Mar 2011, at 16:38, Gordon Henderson wrote:
1.2 has been the most stable version for me.
Same setups with 1.4 +DAHDI has never been as stable with random crashes and
re-starts - however they're not predictable and sometimes months apart. I had
one instance of 1.2 run for over a year
On 24 Mar 2011, at 16:46, tahar .H wrote:
so plz is there any one who can give me a puch to learn this extraordinary
Asterisk plz(video things will be better :))
Learn to ask questions. Learn to read books. Learn to use google.
S
--
On 23 Mar 2011, at 10:40, Nikhil wrote:
I am planning to use asterisk as a IP phone(Porting asterisk into a hardware).
Interesting..
Is there any limitations if I use asterisk as a SIP client?,and asterisk has
any advantages if use like this?
It's not really designed as a SIP client. It's
On 22 Mar 2011, at 01:09, Outback Dingo wrote:
Even worse... now it smells of a scam
At least their website isn't hideous...
Oh..wait.. ;)
S
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On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
I also notice the presence of a Remote-Party-ID SIPheader... Where does
this come from ?! Not from my dialplan...
sendrpid in your sip.conf
Steve--
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On 15 Mar 2011, at 11:30, Jonas Kellens wrote:
On 03/15/2011 12:24 PM, Steven Howes wrote:
On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
I also notice the presence of a Remote-Party-ID SIPheader... Where does
this come from ?! Not from my dialplan...
sendrpid in your sip.conf
On 15 Mar 2011, at 15:21, Jonas Kellens wrote:
On 03/15/2011 12:39 PM, Steven Howes wrote:
On 15 Mar 2011, at 11:30, Jonas Kellens wrote:
On 03/15/2011 12:24 PM, Steven Howes wrote:
On 15 Mar 2011, at 09:08, Jonas Kellens wrote:
I also notice the presence of a Remote-Party-ID SIPheader
On 14 Mar 2011, at 15:58, Jonas Kellens wrote:
dialplan :
exten = 67121212,1,NoOp()
exten = 67121212,n,Set(CALLERID(all)=3259 3259)
exten = 67121212,n,SIPAddHeader(P-Preferred-Identity:
sip:3259\;user=phone)
exten = 67121212,n,SIPAddHeader(Privacy: id)
exten =
On 14 Mar 2011, at 16:24, satish patel wrote:
I test page application and it works but i am worried about i have 200 SIP
phone. Do you think asterisk page application can handle that number of page
?
Do they support multicast?
S--
On 3 Mar 2011, at 20:53, Danny Nicholas wrote:
Not having an in-depth knowledge of how EU numbering works
Sadly there is no 'EU numbering'. Europe isn't a country, thus doesn't share
any dial plan. There appears to be some tendency towards having a '7' at the
front of a mobile number, but it's
On 28 Feb 2011, at 10:33, Rizwan Hisham wrote:
The problem I have been experiencing since last month is that some of my
customers are getting calls with Asterisk Unknown caller id. Most of them
in the middle of the night. And my asterisk server has no record of these
calls. The customers
On 17 Feb 2011, at 10:04, Nikhil wrote:
Do I need to modify chan_phone application to make it works or it is
available in net.
Why not use a proper sip client?
S
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On 24 Oct 2008, at 03:57, David Gibbons wrote:
Dare I ask why you want to do this?
Cheaper than buying an AIM-CUE? And certainly more flexible.
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To
On 22 Oct 2008, at 20:29, Craig Van Ham wrote:
HI all,
snip
This appears to be the same message you posted earlier.
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On 14 Oct 2008, at 18:05, Christian Victor wrote:
Steven Howes schrieb:
Have created a system that involves using call files in the outgoing
spool folder. On some occasions it retries which is fine is there
any way to view calls waiting retries from the CLI? Using 1.4 btw.
Have googled
On 16 Oct 2008, at 14:57, jonathan augenstine wrote:
I am trying to build app_confcall and it is failing. Are there
known build issues with this module. I am running Asterisk 1.6.0-
beta9.
Ah yes. 'failing'. I bet that is all it says eh? its not like
compilers give descriptive errors is
Hi All,
Have created a system that involves using call files in the outgoing
spool folder. On some occasions it retries which is fine is there
any way to view calls waiting retries from the CLI? Using 1.4 btw.
Have googled to no avail (although it is near the end of the day so I
might
On 14 Oct 2008, at 11:00, Chris Rowson wrote:
Hi folks,
I'm working on a solution using the Asterisk voicemail component and
wondered if anyone knew the answer to this question please?
I understand that Asterisk saves voicemail to
/var/spool/asterisk/voicemail/context/mailbox/INBOX/ but I
Hi
triXbox.org can answer these questions. Google may also give a
balanced view. But yes, i can assure you, people are using Trixbox
from Fonality.
Steve
On 6 Oct 2008, at 10:24, broadband Voice wrote:
Anyone using Tribox from Fonality. I understand its open source and
free. Can I use
Just copy the src folder and do `make install` on each machine?
Then tar and copy the /etc/asterisk folder if config is important too.
On 29 Sep 2008, at 08:41, Jim Boykin wrote:
Is there a script to create an Asterisk binary package after it is
compiled on one system.
We do not want to
Hi,
Agreed. Asterisk on a VM appears to work sometimes, only if magic is
involved. It is not the way to run anything for a business.
Steve
On 25 Sep 2008, at 02:36, Dean Collins wrote:
Mike,
Buy an asterisk appliance like http://www.taa.com/products-vdex-40.html
problem solved.
If
On 25 Sep 2008, at 18:38, Shyju K wrote:
I was configuring asterisk with TE110P Card.When run zttool
It is showing Blue Alarm/Yellow Alarm/Recovering and the
card's LED is blinking RED and GREEN.
I have connected 12,45 Lines from ISDN modem(RAD ASMi-52)
to 12,45 of the PRI card
Hi,
We saw this between Asterisk and an Audiocodes gateway. Whilst the
voicemail is being recorded asterisk is not sending *ANY* rtp. Silence
detection will always detect silence if it listens to this side of the
conversation. Adjusting the threshold wont work, you need to find the
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