To be sure you could setup a soft phone and see if the caller ID name comes in
correctly.
On Mar 10, 2015, at 8:41 AM, Jordan Cook - Gyron Networks
jordan.c...@gyron.net wrote:
Hi,
In my dialplan I have the following line.
same = n,Set(CALLERID(name)=Support)
I am expecting
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd R.
Sent: Monday, January 19, 2015 1:45 PM
To: Asterisk-Users List
Subject: Re: [asterisk-users] sip show channelstats reliable? Additional info:
At the moment I am running 1.8.x but the other day I was getting the same
results on 11.x
Additional info:
At the moment I am running 1.8.x but the other day I was getting the same
results on 11.x
Here is a sample from show channelstats. I do think this command is showing
that there is trouble between specific IP's and my Asterisk box but I don't
know if the numbers are accurate and
I am seeing lots of lost packets when running the command sip show channelstats
at the CLI.
There are issues across multiple Asterisk servers I am trying to diagnose but
everything I read seems to point to this command being pretty unreliable.
Can I trust the info this command shows?
I am
Right now we I am using Asterisk boxes as a gateway between our Level 3 SIP
trunks and our customer PBXs.
I love and understand Asterisk but the company I am working for is looking for
a more Commercial type solution where we can go to a vendor for support etc.
I know, we can get Asterisk
variable.
Date: Mon, 27 Oct 2014 08:51:42 -0500
Subject: Re: [asterisk-users] Setting Music on Hold with the Manager Interface
From: mjor...@digium.com
To: tjrl...@live.com; asterisk-users@lists.digium.com
On Sun, Oct 26, 2014 at 10:42 PM, Todd R. tjrl...@live.com wrote:
Does anyone know how
and didn't seem right anyhow
since it already knows it's a channel variable.
Thanks in advance for any help on this.# Set the Music on Hold
fputs($socket2, Action: Setvar\r\n);
fputs($socket2, Channel: .$channel.\r\n);
fputs($socket2, Variable: musicclass\r\n);
fputs($socket2, Value: .$mohclass.\r\n
OK, been messing with Asterisk for a long time and I have my opinion on where
the issues lies but sometimes it's just nice to see what others think that can
relate :-)
Here goes..
Inbound calls flow like this:Tier 1 Provider (SIP) Asterisk 1.8 Name Brand
PBX - Calls work fine
Outbound calls
Hi, all.
Is there any chance to set individual CALLERID(num) for channels SIP/peer1,
SIP/peer2 in a call Dial(SIP/peer1SIP/peer2). There is an option to use
Dial(SIP/peer1SIP/peer2,,M(set_callerid)), but the macro will be launched
after the channel answered. Not really want to use local
Been working with Asterisk for a long time but this is the first time I have
dealt with this issue.
I am setting up an Asterisk box (FreePBX not my choice) to interface with an
e911 provider.
They say their switches only listen for RTP on ports 2-21001 which is
outside the normal range
Search google for Asterisk Manager Interface.
For the most part, if you have raw Asterisk installed then that's what you get
and have to build what you want on top of it or hire a developer to do it.
Date: Fri, 10 Jan 2014 12:12:47 -0500
From: szilvertho...@gmail.com
To:
Luminvox is one.. There are others out there..
Here's an article by Ward Mundy that might help:http://nerdvittles.com/?p=7448
From: jpra...@gmail.com
Date: Fri, 10 Jan 2014 12:16:43 -0800
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Text to Speech Engine
Hello,
Anyone know
Actually, scratch that.. Luminvox is not text to speech it's speech recognition
software. Got this mixed up and turned around :-) Anyhow, see the link I posted
earlier, it's got some good info to get you started.
From: tjrl...@live.com
To: asterisk-users@lists.digium.com
Date: Fri, 10 Jan 2014
May not be what you are looking for exactly but search Google for Nerd Vittles
BeagleBone. I am not suggesting you use that exact solution but, reading the
article with give you all sorts if ideas about what you could use in your
situation.
The BeagleBone is a small form factor computer like
I do this by writing custom CDR. I write the agents extension write into the
CDR records. This makes is easy to just parse through the CDR and get all the
info you need about the call.
Google something like asterisk custom CDR
On Nov 29, 2013, at 11:36 AM, Leandro Dardini
Did you have the externalip setting in sip.conf set to the Elastic IP?
Date: Sat, 23 Nov 2013 23:42:36 -0500
From: ja...@fivecats.org
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby
system?
On 11/22/2013 12:52 PM, Todd R
Just checking one more time to see if anyone has an opinion on this. I am
primarily interested in using a cloud type setup such as Amazon AWS for the
redundancy, easy backup and recovery options. It's not about price but the idea
that it will be very hard for a single piece of hardware to ruin
thinktwice about Amazon -- and virtual in general is not a good
idea for this sort of thing. I have seen messages about bad results
with amazon specifically.
Todd R. tjrl...@live.com wrote:
Just checking one more time to see if anyone has an opinion on this. I am
primarily interested
.
Todd R. tjrl...@live.com wrote:
Just checking one more time to see if anyone has an opinion on this. I am
primarily interested in using a cloud type setup such as Amazon AWS for the
redundancy, easy backup and recovery options. It's not about price but the
idea that it will be very
Took me a while but I have finally embraced cloud computing and all the
benefits.
The only thing I have yet to feel comfortable about putting in the cloud is
real live Asterisk boxes to be used in production. I know it's being done
because as far as I know Twilio is using Amazon for their
What do you want to happen once the call is made?
You can choose to fire the call off using the originate command with the
Asterisk Manager Interface from a PHP page or some other similar language. No
need for Perl on the Asterisk box at all really unless you need it for
something else.
Well, first of all, my name is Ezequiel and I'd been on this list for a
very short time, but I see a lot of people willing to help here, so I'll
give my problem a try here.
After using asterisknow for almost a year, I decided to give plain
asterisk a try, so I installed CentOS 6.4 and Asterisk
Just put a new phone in place with the latest firmware from Cisco. This is the
first SPA501G we have with this firmware.
In the Asterisk CLI we are now seeing the error message below about once every
second. When we unplug the phone, the messages quit.
NOTICE[15539]: chan_sip.c:9602
Trying to figure out the best way to pull an active call out of a queue by
unique id and put it on hold. I don't want to put it on hold on the agent's
phone but I want it to be pulled away from the agent's phone and into Asterisk
limbo somewhere.
Shortly after I want to pull the same call out
I currently have all agents/members logged in with local channels. When a call
is sent to one of the agents, then the agent transfers the call out the line
frees up on their phone but still shows in-use until the call that was
transferred is hung up.
How can I free up the agent/local channel
finally found the StateInterface: option available in 1.6
and above. I added it to my PHP login screen like this..
fputs($socket2, StateInterface: SIP/.$agentid.\r\n);
The problem is that the queue was monitoring the local channel in terms of when
a call was hungup or not, allowing other calls to come
Hello List,
In PHPAGI, I'm using the Astrisk Manager function send_request() to
originate an outbound call. I want to execute the remaining PHP code after
the call gets executed (depending on user input). But presently the call
originates in a different context and asterisk executes the remaining
Gracias
2012/7/14 rnew...@digium.com
This message was sent to test a problem with the mailing list.
Please ignore it, and we apologize for any inconvenience.
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Hi y'all,
is there a possibility from within asterisk to write CDR into can
ical-calendar? Being a (part time) IT-Support-Monkey for almost two decades
and probably brain-dead since the late 90ties, i usually can't remember to
whom i've talked to yesterday, let alone a week ago ;). If i could see
Hiya,
SIP Messaging is implemented in asterisk-10...
The only documentation I can find talks about a patch and is pretty
old:http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging
http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging
Like anything on voip-info.org it's horrible
Hiya,
can i use an XMPP Client to see the presence of a hint? I have configured
asterisk in component-mode, seem's to work, but all users (
xmpp:1...@asterisk.dohmain.com are online, even if 123 isn't a configured
hint). Any good howto's out there, all the stuff on voip-info.org is
completely
I've been toying around with the idea of starting some kind of 'Open
CNAM' project to destroy the current money hustle BS that dominates this
industry. The ever-growing FreeCNAM database may be a good starting
point for such a project.
I would also like to use Bitcoin (BTC) as the
FreeCNAM.org is providing a free CNAM API for Open Source PBX users.
This API queries a private CNAM database, and returns standard
15-Character CNAM results. Any entry not already in the database will
be queued for investigation, and added to the database as soon as
information is located. This
has any thing to do with it but... something is
fishy here.
Cary
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael R.
Wally
Sent: Sunday, May 29, 2011 6:48 AM
To: asterisk-users@lists.digium.com
Subject
Try them all again. Remember that this is a static database that has to
'research' numbers it has not seen before.
By now (a few minutes later), the database should have been updated.
On 05/29/2011 08:56 AM, Richard Kenner wrote:
FreeCNAM.org is providing a free CNAM API for Open Source PBX
The system uses real Telco CNAM Dips. Any generic names you get are
from the subscriber's carrier itself. We can only provide what we
ourselves get.
I tried it, but it returns the same kind of junk that some of the databases
do. For example, on a Florida number, it just says FLORIDA instead
Hi,
i have some spare (read: Boss get's a new one every few month ;)) Android
Phones laying around. Does someone know a way of using them as a mobile
gateway for asterisk? I could not find any SIP-Gateway in the Market, and i
don't think it's possible to use the GSM Audio directly with something
to Tim Panton and PhoneFromHere.com
PSTN: (567) 252-2286 thanks to Alex Graham Bell
iNum: +883 5100 123 94882
Text on IRC #vuc on Freenode.net - http://vuc.me/irc
If in doubt about the time in your zone, look here: http://vuc.me/next
Hear you there...
/r
: +883510001826724
Join us and contribute your knowledge and experience or learn from others.
The VUC guarantee: There is never any top posting on our conferences!
/r
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that are
safe all over the USA? Is there a check of area codes?
Color me just curious...
/r
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://vuc.li/mp3stream
None of the above will work until about 15 minutes before. The mp3
stream goes live on the hour.
The VUC may be the only way to hear John Todd's voice at the moment.
That alone is worth the price of admission!
/r
:30 and 1PM EST Friday. For local time: http://vuc.me/next
To connect with the VUC, one address for all the info: http://vuc.me
Hope to see youthetre.
/r
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-906-3916
iNum: +883 5100 123 94882
IRC: #vuc on Freenode.net or http://vuc.me/irc
VUC in your local time zone: http://vuc.me/next
Also, we are giving away a Gigaset SIP/DECT phone that day, so relax,
join us, mute if you need to belch after all that food and drink and
get ready for IPv6!
/r
...@login.zipdx.com in G722
wideband or see http://vuc.me for more info about
Hope to hear you there?
/r
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#vuc on Freenode.net
Hear you there?
/r
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-- Forwarded message --
From: Peter Kunz munged
Been there, many, many times.
http://xkcd.com/806/
Look at this comic, you will laugh, I guarantee it!
Thanks Peter!
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Hear you there!
/r
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hours.
See you there!
/r
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asterisk-users
to another number?
See Dialplan.
Thanks for any advice you might have. I'm very new to all of this...
Take the first advice, look for a book, there are many but Asterisk
The Future of Telephony is excellent and you can get everywhere.
/r
On the S675IP SMS is here:
Messaging - SMS - Settings
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on fixed. There is nothing on the web menu, only the
handset menus.
/r
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gigaset
script and I'll bet you will find something. I know someone did this
(I was going to use it but changed my mind).
r
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service from France Télécom, that's the way it worked.
/r
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for them to stop it (well documented here in the archives
by myself and others)
Amazon did something about it? I don't remember seeing that, Gordon,
it's a new record. The average response has been zero.
/r
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, Amazon will be
put in that position. Until then, they have done very little, and we
have stopped using their cloud services as much as possible.
/r
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New
Hi,
i'm trying to get PoC on Nokia Phones to work with asterisk. I think the
store-and-forward part could easy be done in the dialplan, but i can't even
get the handset to register with asterisk (authentication failed). I'd try'd
to find the difference between pure sip and PoC-SIP, but didn't
here: http://openbts.sourceforge.net/NiuePilot/
and more about the installation here:
http://vuc.me/2010/island-telephony-adventure/
/r
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anytime on Freenode.net #vuc
/r
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asterisk
so special.
That's YOU!
/r
ps: Allison is at her top form too. Make sure you follow Allison on
Twitter, she's @voicegal
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We should be hearing more on this from Darren either this Friday or next on VUC.
http://vuc.me
/r
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and simplify the
expressions, then play with the regex tests.
/r
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On Fri, Jul 30, 2010 at 11:54 AM, Alex Bell voicese...@gmail.com wrote:
/r,
r u not on talkshoe anymore? This is 2 weeks in a row that I've clicked
in, but no one was home? At least last week I was one of 2 guests, today I
was all by my lonesome... :(
Hi Alex,
When I'm not in my own
on Freenode.net or http://vuc.me/irc
Info about VUC is htp://vuc.me
Best,
/r
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Hi,
Since f2b is one of the topics du jour here, I was wondering if
someone would mind telling me what these pf stats mean:
Evaluations: 964303 Packets: 12176 Bytes: 648408 States: 0
Looks like pf examined nearly a million cases from fail2ban in 24h?
thanks,
/r
On Wed, Jul 28, 2010 at 9:03 AM, Kyle Kienapfel doctor.w...@gmail.com wrote:
On Wed, Jul 28, 2010 at 6:38 AM, Randy R randulo2...@gmail.com wrote:
Hi,
Since f2b is one of the topics du jour here, I was wondering if
someone would mind telling me what these pf stats mean:
Evaluations: 964303
as anyone and he con
help you find someone. Kevin's advice is good (as usual): check out
Adhearsion. Jason is a part of that, too.
/r
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and
particularly their answer to complaints, which was something like
Deal with it.
/r
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On Sun, Jul 25, 2010 at 10:20 PM, Janu Mukherjee janu.mu...@gmail.com wrote:
I installed asterisk server in my linux box. I configured a user 1000 using
xlite and registered with asterisk server in the same linux box. I
Where on the network is this box?
configured one more user 1001 in other
…). It seems to me
that a quick fix would be to have the system restart fail2ban whenever the
firewall is restarted. Has anyone else encountered this issue? …and come
up with a solution?
I believe there's a way to make the rules persist in a file. (see the
fail2ban docs)
/r
On Mon, Jul 26, 2010 at 12:19 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Why isn't the Asterisk box on a static IP on the LAN? That seems to be
asking for trouble using DHCP.
I was assuming he meant the ISP DHCP renewal.
/r
there?
/r
Get 20% off of Astricon ticket with the VUC discount code - I'll give
it at the very beginning today and you can ask in IRC if you come to
the conference late.
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.
That way, the users can also change their configs to 587 and most
spammers will be trying 25 which is closed.
Is this a tenable idea? What are your experiences and opinions?
tia
/r
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familiar with Google's pro services, we have
other customers on them. However, this doesn't address the question I
asked, which is regarding a FreeBSD mail server we run.
/r
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spam, I get some which gets by the filters.
Hi John,
I'd like to do that, but there are nomad users who might be anywhere
in the world. True maybe I could ask them to use port 587 and then
allow ONLY the service IPs access to port 25.
/r
On Tue, Jul 13, 2010 at 12:53 PM, cov...@ccs.covici.com wrote:
What you can do -- I don't know about nomad, but can you make them use
authentication?
They do identify, but they have to connect first :)
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On Tue, Jul 13, 2010 at 12:58 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Tuesday 13 Jul 2010, Randy R wrote:
I was thinking of closing port 25 and using an alternate port (587?)
setup if the spam service is able to connect to an alternate port.
That way, the users can also change
before I read your message) I decided to limit port 25 access
to the restricted IP set we know about. This will be an interesting 48
hours or so while we see if the users are still using port 25 :-)
/r
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. I don't know how many IP can be put in a pf table but it was
obvious that there would be tens of thousands in a very small number
of days.
/r
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New
that no one has written a more modern version of something
like Teraterm, but maybe the majority of Windows users don't do SSH?
The fact is that when I mention SFTP to them (we don't do ftp at all
usually) I can hear the crickets over the phone.
/r
PS: http://www.ayera.com/teraterm/
I'm pretty sure there was a last update or patch or something because
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we only use ssh2 on our servers. It's been at least 2 years since I
used Windows for SSH though.
What's wrong with putty?
Nothing's wrong with it, I just didn't like it as well.
/r
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to remember to look. It's
been a long time since I've used it for daily work.
/r
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of being a VUC regular, Michael
Iedema of Askozia embedded pbx met Randal Schwartz at Astricon last
year and this week will be a guest on FLOSS Weekly July 7th.
http://twit.tv/FLOSS
/r
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Hi Guy's,
i'm having a wheird problem with an asterisk-trunk installation, dahdi via
hfc-s sounds extremely distorted, but the same installation (same libpri,
same dahdi etc) is ok with asterisk 1.4. I dont see any errors, neighter on
the console nor the error logs. Any ideas what was changed?
of SfS !
/r
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asterisk-users mailing list
There are some good suggestions here as a starting point:
http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/
Rgds,
mcr
On 29 June 2010 15:39, Rodrigo Lang rodrigoferreiral...@gmail.com wrote:
Hello list.
I'm trying to find a way to block any ip that tries to login more
Zaptel and Asterisk.
IMO, if it's a business phone, you'd do well to just reboot it at 3AM
once a week or once a month or some interval that you're comfortable
with. We used to do this for a similar reason.
/r
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to add the larger images to that.
/r
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On Fri, Jun 25, 2010 at 11:44 AM, Gilles codecompl...@free.fr wrote:
On Fri, 25 Jun 2010 09:53:34 +0200, Randy R randulo2...@gmail.com
once a week or once a month or some interval that you're comfortable
with. We used to do this for a similar reason.
Right, but he won't remember to do
On Fri, Jun 25, 2010 at 12:10 PM, Gilles codecompl...@free.fr wrote:
Sorry for the misunderstanding. So I can just run reboot from a CRON
job then.
From root's cron, yes
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See you there.
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is that magic number?
Thanks in advance for any ideas.
/r
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http://www.engadget.com/2010/06/22/skypekit-beta-sdk-adds-skype-to-any-application-or-device/
Great! Finally a change to get a chan_skype without beeing a**-raped by the
copyprotection (which is the sole reason i didn't buy it), and maybe even
more than the absolute basic features (like Silk and
://vuc.me for more
ways to connect.
At 1PM EDT, Counterpath will join us to talk about their new Bria iPhone app.
/r
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. $5 for the channel and $10 for Skype Manager.
Maybe something for each name, too?
/r
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as I mentioned above. In addition to the binary, youneed
to pay for Skype Manager and each seat on that (name) - at least that
is my understand of their page.
/r
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On Tue, Jun 15, 2010 at 9:46 AM, Thomas Kenyon
dig...@sanguinarius.co.uk wrote:
On 15/6/10 06:22, Randy R wrote:
In October, they will begin charging for Skype Manager (required for
SfS) and a per seat charge for that.
SfA also requires Skype Manager, and only works with users that were
on this - so this means x86, right?
Recommendations wanted
* What hardware
* What distro
* Which Asterisk version
Comments and suggestions welcome. This is going to be discussed on VUC
as well, so if you're comfortable with it, come on by: http://vuc.me
Thanks in advance,
/r
On Mon, Jun 14, 2010 at 1:37 PM, Chris Bagnall
aster...@lists.minotaur.cc wrote:
We've used the Asus eeeBox (desktop version of their little netbooks) quite
successfully in past projects: Atom 1.6, 1GB RAM, 160GB HDD.
Wow, we used to benefit from the space program that handed down
technologies
good stuff snipped
Thanks everyone for your suggestions.
Is it feasable to run Skype for Asterisk on the Atom processors? It's
a feature I'd really like to have. As for conferencing, we rarely use
it but never would need more than 3 seats.
for you top posters, the good stuff would have been
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