Hi Everyone,
We are using Queuemetrics but it doesn't Record the Hold Time as it's never
logged on the queue_log file. However, when an agent or an extension presses
HOLD button on their phone, asterisk does create an event for Music On Hold
which is logged in the /var/log/asterisk/full.
I want
Hi Everyone,
We use the top buttons on Aastra 55i to login and logout from Queues. This
is the order:
Button 1 = Login to English Queue
Button 2 = Login to Spanish Queue
Button 3 = Logout of English/Spanish Queues
There are indicator LEDs on each of these buttons. Is there anyway we can
send a
Amazing. Thank you very much.
Unfortunately, the phone type is 53i and not the 55i as I mistakenly noted.
It has only 6 buttons on the left side. Is there a workaround for this?
Thanks again.
-Bruce
On Wed, Oct 20, 2010 at 5:12 PM, bakko asannu...@gmail.com wrote:
Hello,
you can't utilice
Here is the login for English:
;English-temp LOGIN
exten = 800,1,Answer()
exten = 800,n,AddQueueMember(500|Local/${CALLERID(num)}...@from-internal/n)
exten = 800,n,Set(DEVSTATE(Custom:agenten)=INUSE)
exten = 800,n,Playback(agent-loginok)
exten = 800,n,Hangup()
;English Logout
;All Queues Logout
Thanks for the input. By this configuartion you mean by the way I do Add and
Remove member from the Queue?
Can you please explain by what sort of configuration (what to use instead of
Add and Remove queue member) would get this working.
I guess I am looking for speedial/BLF on the same key ?!!!
Hi Everyone,
For some reason a few phones connected to a pfSense box can't make or allow
in OpenVPN in port 1194 UDP. So, I established the VPN tunnel on 1194 TCP
and it works fine. I would like to know if there is any disadvantages to
using TCP over UDP for the tunnel when using Asterisk or is
If you want to turn off the audio totally you can set audio to NO (it's
probably the 4th or 5th in list of Global settings). Otherway is to blank
the file responsible to play that file and keeping the settings intact.
However, there are numerous options to turn on and off the various
announcements
Anything on this guys?
I am sure someone had the need to record the HOLD time or maybe it is
already being recorded somewhere?
Any thoughts are appreciated.
Thanks,
Bruce
On Wed, Oct 20, 2010 at 3:30 AM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
We are using Queuemetrics
, Oct 25, 2010 at 4:51 AM, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
I would probably do this through the AMI, it should spew out the info you
require. Timestamp when they entered the queue and timestamp when they get
answered.
On 10/25/2010 05:01 AM, Bruce B wrote:
Anything
Of *Bruce B
*Sent:* Monday, October 25, 2010 9:32 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Best way to recording the hold time for a
Queue agent or an extension
Thanks for the feedback. I don't need the Queue times but rather putting
Hi Everyone,
Which paid or unpaid commercial plugin is available out there for Asterisk
that would do Outlook contacts pop-up that is proven to work great with MS
Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well
through the Outlook.
Thanks,
Bruce
--
:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B
*Sent:* Monday, October 25, 2010 1:14 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Pop-up for MS Outlook 2007
Hi Everyone,
Just noted that PBXinaFLASH failed me again on yet something else. The
uniqueid field didn't update on the asteriskcdrdb in the past few months but
it is available in the .csv files in /var/log/asterisk/cdr-csv/*.csv
I have already re-did the asterisk-addons install with correct
Hi Everyone,
I am doing a queue show and I can't see any column that shows a queue member
priority. Is there any other command that can show the member priority on
the Asterisk 1.4x CLI?
We are using this format of dialplan to login agents:
exten = 123,Answer()
exten =
Thanks Warren. That should do.
Regards,
Bruce
On Thu, Nov 4, 2010 at 2:54 PM, Warren Selby wcse...@selbytech.com wrote:
On Thu, Nov 4, 2010 at 12:56 PM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am doing a queue show and I can't see any column that shows a queue
member priority
Hi Everyone,
We have three different Queues set to leastrecent strategy and from time
to time I hear someone complain that they receive short rings (partial ring
cycle) and since it's not their turn even if they pickup the phone the call
is not given to them since the Queue is actually hitting
,
Bruce
On Thu, Nov 4, 2010 at 9:29 PM, Chad Wallace cwall...@lodgingcompany.comwrote:
On Thu, 4 Nov 2010 20:12:54 -0400
Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
We have three different Queues set to leastrecent strategy and from
time to time I hear someone complain
Yeah, I think I had it set to 2 seconds and that creates that short ring on
another extension.
Thanks,
On Fri, Nov 5, 2010 at 9:47 AM, Mark Deneen mden...@gmail.com wrote:
On Fri, Nov 5, 2010 at 1:18 AM, Bruce B bruceb...@gmail.com wrote:
Chad,
You are absolutely right on this one. I had
seconds is just three rings complete.
Thanks,
Bruce
On Fri, Nov 5, 2010 at 11:31 AM, Mark Deneen mden...@gmail.com wrote:
On Fri, Nov 5, 2010 at 10:38 AM, Bruce B bruceb...@gmail.com wrote:
Yeah, I think I had it set to 2 seconds and that creates that short ring
on
another extension.
Thanks
Hi Everyone,
Configuring a Polycom conference bridge IP 5000 to connect to Asterisk. For
some reason I don't see any SIP packets coming in to Asterisk at all. I
don't want to use XML or ftp etc for now and just use the Web Interface to
get it going with basic features. But the Web UI is a bit
Hi Everyone,
Is there other comparable products to Proxmox to be used for Asterisk
instances? Ease of use, web interface, and Asterisk/CentOS support would be
ideal.
Thanks
--
_
-- Bandwidth and Colocation Provided by
Thanks. OpenNode seems promising and neat. Proxmox is disappointing when it
comes to their forums and documentation. Only few videos listedanyhow.
OpenNode is promising but doesn't have a Web UI yet. Is there anything else
as well?
Thanks
On Fri, Nov 5, 2010 at 4:06 PM, Tim Nelson
Hi Everyone,
Knowing that running Asterisk on an embedded board like the Alix2d3 requires
some fine tuning. Do you know of any good guides out there that does this
from beginning to end? Looking to run this in a small office environment.
Thanks
--
be production ready as well. Meaning solid, reliable
machine.
Thanks
On Sun, Nov 7, 2010 at 12:28 PM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Sun, Nov 7, 2010 at 11:23 AM, Bruce B bruceb...@gmail.com wrote:
Knowing that running Asterisk on an embedded board like the Alix2d3
Thanks for the input. I am looking to use it as a DHCP server as well. And I
also I want it as a VPN server so that I can securely log in to it from time
to time to monitor it's state.
The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk).
Wondering if those two service would play
08:34 AM, Bruce B wrote:
Thanks for the input. I am looking to use it as a DHCP server as well.
And I also I want it as a VPN server so that I can securely log in to it
from time to time to monitor it's state.
The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk
.
Thanks
On Mon, Nov 8, 2010 at 7:24 PM, John Novack
jnov...@stromberg-carlson.orgwrote:
Bruce B wrote:
Thanks. I think I would still need a firewall. Maybe a 1u rack
double enclosure for two Alix boards - one as firewall - and one as PBX
would do better.
Anyhow, I don't want to open the box
Hi Everyone,
I have pfSense running which supplies Asterisk with DHCP. I had some testing
ports opened for a web server which I have totally closed now but when I
chose option 10 (filter log) on pfSense I get all of this type of traffic
(note that it was only 1 single IP and once I blocked that
wrote:
Bruce B wrote:
Hi Everyone,
I have pfSense running which supplies Asterisk with DHCP. I had some
testing ports opened for a web server which I have totally closed now but
when I chose option 10 (filter log) on pfSense I get all of this type of
traffic (note that it was only 1
, Bruce B wrote:
Yes, it is a small office. I am familiar with pfSense. I am not sure if
firewall on Astlinux is as versatile and flexible. But also, I am
wondering
if with all those attacks around now-a-days if the box will be able to
handle 5 extensions, voicemail, IVR, firewall, DHCP
Hi Everyone,
I don't have much experience with eSXI. I can really use some advise on how
to run it without any trouble with Asterisk on CentOS VMs.
First of all, is it a good option to run multiple hosted Asterisk instances
on a VMware eSXI? or would you rather prefer something like Xen,
Hi Everyone,
I am looking into AMI (using PHP) to record every instance of HOLD that is
generated by putting a caller on HOLD (press hold button on the phone set).
There is no HOLD in Asterisk but the event Music on Hold is generated when
HOLD is pressed. The complexity is that all of the the
Hi Everyone,
I am wondering why documentation of some of the vital parts of Asterisk is
hosted on voipinfo.org (unreliable is some parts) and not on asterisk.org?
For example the list of AMI events are not well documented and one has to
guess which version supports which event. The documentation
To be honest this is the first time I see this wiki mentioned. It doesn't
even come up in talks on this list. The wiki should be advertised often and
there should be some sort of active monitoring and supervision of the
contents as well as some serious ongoing official contributions. All this
well
Hi Everyone,
There are situations when internet connection is lost, SIP provider fails,
or even authentication to SIP provider fails, and we want to use the backup
Dahdi channels (PSTN). As simple as it may sound but with the
many different situations and error messages it seems like it's not so
Thanks for the input guys. I really appreciate all the input and I am sure
they work but I thought there would be a much better way to do this. Sounds
like patching things to me. Why doesn't Asterisk take advantage of the
qualify values to make sure if the SIP connection is up or not? Shouldn't
Hi Everyone,
I am using pfSense to do firewall and NAT on an Asterisk server. I have
ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local IP
192.168.5.5. However, when a user from outside using Linksys WRP400 ata
connects to the Asterisk server and registers I see them as
AM, Ryan Wagoner rswago...@gmail.com wrote:
On Sat, Dec 11, 2010 at 3:06 AM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am using pfSense to do firewall and NAT on an Asterisk server. I have
ports 5060 TCP/UDP and 10k-20k UDP forwarded to the Asterisk server local
IP
192.168.5.5
Hi Wang,
Did you mean to write a feedback? You sent an empty message.
Regards,
On Sat, Dec 11, 2010 at 11:56 AM, w...@pythian.com wrote:
Sent from my “contract free” BlackBerry® smartphone on the WIND network.
-Original Message-
From: Bruce B bruceb...@gmail.com
Sender: asterisk
**
Regards,
Bruce
On Sat, Dec 11, 2010 at 10:15 AM, Ryan Wagoner rswago...@gmail.com wrote:
On Sat, Dec 11, 2010 at 3:06 AM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am using pfSense to do firewall and NAT on an Asterisk server. I have
:
On Sat, Dec 11, 2010 at 11:45 AM, Bruce B bruceb...@gmail.com wrote:
Thanks for the feedback Ryan.
Siproxd is not installed. I think Siproxd like you said just does the
reverse meaning if phones are part of pfSense subnet then it connects to
outside world. But in my case they are coming
Hi Everyone,
I ocassionally hear echo, static, and garbled voice when calling extension
to extension between two office (different geographic locations connected
using OpenVPN - 1 with DSL and other with T1 - 1500 KM apart). I am guessing
it's a bandwidth or jitter issue that is giving me faint
Nortel 1535. Does video as well.
On Fri, Dec 17, 2010 at 10:40 AM, Matt mhop...@gmail.com wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
--
_
-- Bandwidth and Colocation Provided by
Hi Everyone,
I am trying to install the new cdr-stats from http://www.cdr-stats.org/ for
Asterisk 1.6 but it's installation instructions are all garbled. It mentions
both sqlite and mysql and there are no organized documentation. Not to
mention that the apache port 8000 and port 9000 are also
Hi Everyone,
I understand that there are a few warnings about using cp to move .call
files to /var/spool/asterisk/outgoing as they might acted upon before copy
is done. So, using PHP, What is the equivalent of mv command? Would it be
rename() in php or is there a better method?
Thanks,
--
This is a NAT issue like noted before.
Try:
localnet=192.168.0.0/ http://192.168.0.0/24255.255.255.0
instead of:
localnet=192.168.0.0/24
http://192.168.0.0/24Also, make sure you have all your VPN connections as
localnet and other side subnet as localnet as well if you are using VPN.
Otherwise,
Thanks for looking into it. Yes, it missed up and not worth looking at it.
Unfortuantly, so are a few products from the same company (probably trying
to make money of support which I understand)but it seems they released
an install script which is here for CentOS:
Hi Everyone,
I use Asterisk for regularPBX use it's made for. But I want to take it a bit
further and use it at cmmand level to be able to send SIP notifies to
restart a phone or take advantage of a phone's UPnP capabilities. Is
Asterisk capable of that? If so, what is a simple SIP reboot message
Thanks Kai-Uwe and everyone else. I have seen all those examples and I am
exploring the sip_notify.conf file now which makes things more clear to me.
However, when sending a SIP notify to a phone that is not registered to
Asterisk via it's IP address should I expect to receive a success of fail
Hi Everyone,
I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can
originate calls see the program login nicely but when a call comes in it
only shows the Name portion of the CLID and not the number hence it pulls up
a new contact on Outlook. The new contact only show name
Thanks for feedback. I am looking mainly for pop-up of Outlook and don't
need outgoing call at all but it would be nice to have.
Regards,
On Tue, Dec 28, 2010 at 4:01 AM, Stefan Schmidt s...@sil.at wrote:
Am 28.12.10 07:26, schrieb Bruce B:
Hi Everyone,
I am using OutCall 1.6 (latest
forward to your analysis.
Regards,
Bruce
On Tue, Dec 28, 2010 at 3:58 PM, Moises Silva moises.si...@gmail.comwrote:
On Tue, Dec 28, 2010 at 11:33 AM, Bruce B bruceb...@gmail.com wrote:
I appreciate your feedback and let me know what info I can post here that
may help resolve the issue
-t.co.uk wrote:
Hi Bruce,
On 12/28/2010 10:51 PM, Bruce B wrote:
Thanks for the input. I can not replicate the situation as it happens
randomely or maybe over the weekend. However I have sent you all the
requested command and logs in a separate e-mail for your analyzes. The
only thing that stood
Hi Everyone,
1- Are the Siren7 and Siren14 the G.722 HD voice codecs?
2- Are these codecs only for Polycom units or are they universal across all
other SIP phones that advertise the HD voice codec like Aastra?
3- What is the main difference between the two and is it advisable to run
these over
Hi Guys,
What is out there other than OutCall that works with M$ Outlook and Asterisk
1.4/1.6 ? I prefer opensource and free (as in free in price) but can
consider low price - working - programs as well.
OutCall is giving issues with various versions of Outlook and it always
brings up NEW
wrote:
Hi BB,
you could try this:
http://asterisk-outlook-dialer.voip-singapore.qarchive.org/
Never tested it deeply but apparently seems to work fine.
Giorgio Incantalupo
Bruce B wrote:
Hi Guys,
What is out there other than OutCall that works with M$ Outlook and
Asterisk 1.4/1.6 ? I
Hi Everyone,
I want to know each and every parameter's detail that can be included in
the
read=
write=
in manager.conf
Where can I find this?
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Thanks Paul. That is exactly what I was looking for.
On Sat, Jan 8, 2011 at 2:07 PM, Paul Belanger pabelan...@digium.com wrote:
On 11-01-07 01:33 PM, Bruce B wrote:
Where can I find this?
manager.conf.sample?
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC
, Jan 8, 2011 at 11:27 AM, Steve Edwards asterisk@sedwards.comwrote:
On Fri, 7 Jan 2011, Bruce B wrote:
I want to know each and every parameter's detail that can be included in
the
read=
write=
in manager.conf
Where can I find this?
0) Try and spell check the subject a bit better
Hi Everyone,
I am running multiple instances of Asterisk in Proxmox and so far I had one
central Asterisk feeding all others with trunks from one provider. Now, I
want to connect each Asterisk server directly to the provider. Based on my
understanding, each connection made to the provider port
Hi,
I have OpenVPN and Asterisk working nicely. However, I do use certificates.
Though, it shouldn't matter. Can you explain what doesn't work for you? Is
the connection not established or is the Asterisk and it's client not
communicating?
-Bruce
On Tue, Jan 11, 2011 at 9:20 AM, Gilles
/Application-level_gateway
With kind regards,
Pan
*From:* Bruce B bruceb...@gmail.com
*Sent:* Tuesday, January 11, 2011 8:58 AM
*To:* Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
*Subject:* [asterisk-users] Do I need a sip proxy?
Hi Everyone,
I am
wrote:
On Tue, 11 Jan 2011 10:23:18 -0500, Bruce B bruceb...@gmail.com
wrote:
I have OpenVPN and Asterisk working nicely. However, I do use
certificates.
Though, it shouldn't matter. Can you explain what doesn't work for you? Is
the connection not established or is the Asterisk and it's client
Hi Everyone,
I am looking for a paid version of a program that has proven to work with
Outlook 2007 and Asterisk 1.6 on Windows Vista, XP, and maybe even Windows
7.
Outcall is not the answer as it has lots of bugs and doesn't work.
Something simple with very simple interface would be preferred.
In sip_nat.conf you need to specify 10.8.0.1/24 as your localnet and also
make sure you have your externip setup as well. Else you will notice one way
audio or cut off after 30 seconds. Rest of your work is all good. For
security reasons the workstation that creates the keys is not connected to
As I said, your tunnel address should be part of localnet. Otherwise you
experience what you did.
-Bruce
On Thu, Jan 13, 2011 at 10:00 AM, Gilles codecompl...@free.fr wrote:
On Thu, 13 Jan 2011 15:55:10 +0100, Gilles codecompl...@free.fr
wrote:
The only issue I notice, is that Asterisk
What you need already exists:
http://bestof.nerdvittles.com/applications/screenpop/
http://bestof.nerdvittles.com/applications/screenpop/But better thing
would be to a have TAPI for outlook to query Outlook contact as well because
it allows for making notes on the contact. I am willing to pay
with Inbound and pulls up Outlook contact. Haven't tried outbound.
On Fri, Jan 14, 2011 at 9:19 AM, Gilles codecompl...@free.fr wrote:
On Thu, 13 Jan 2011 17:59:10 -0500, Bruce B bruceb...@gmail.com
wrote:
http://bestof.nerdvittles.com/applications/screenpop/But better thing
would
Hi Everyone,
I am just tweaking a pfSense router and learning lots about NAT etcI
noticed that each call uses four UDP port for RTP. Here is an example of
port for a call I made:
10200
10201
10504
10505
Seems like they are random in pair. I have a restriction of 1-11000 in
my rtp.conf
I mean part of RTP RFC?
On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am just tweaking a pfSense router and learning lots about NAT etcI
noticed that each call uses four UDP port for RTP. Here is an example of
port for a call I made:
10200
10201
On Fri, Jan 14, 2011 at 12:44 PM, Bruce B bruceb...@gmail.com wrote:
I mean part of RTP RFC?
On Fri, Jan 14, 2011 at 2:41 PM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am just tweaking a pfSense router and learning lots about NAT etcI
noticed that each call uses four UDP
Also, to get the SIP very well as well, SIP uses both TCP/UDP 5060 right?
and why are there recommendations of opening 5000-5082 UDP for SIP along
with 5060 TCP? Are there any niceties to that as well? maybe video
transmission stuff?
Thanks again,
On Fri, Jan 14, 2011 at 4:12 PM, Bruce B bruceb
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B
*Sent:* Friday, January 14, 2011 2:15 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Why are 4 ports used for a single call?
Thanks guys. I am not sure whether that call
Bruce B wrote:
Off topic - what is top post? I am using gmail + chrome - no ugly
Outlook.
http://www.justfuckinggoogleit.com/search.pl?query=top+posting
It's why most of the experts in here ignore your posts. If you haven't got
the good sense to follow etiquette, the Delete key becomes
are there recommendations of opening
5000-5082 UDP for SIP along with 5060 TCP? Are there any niceties to that
as well? maybe video transmission stuff?
Thanks
On Fri, Jan 14, 2011 at 6:32 PM, Tom Rymes try...@rymes.com wrote:
On Jan 14, 2011, at 5:24 PM, Bruce B wrote:
So, simply pressing Reply and typing
Hi Everyone,
Are there any generally accepted and widely used tools made and tailored to
be used for purpose of monitoring Asterisk servers? I am wondering if there
is anything that the Asterisk community mostly uses or are there lots of
manual scripts written and nothing really exists that every
On Fri, Jan 14, 2011 at 6:53 PM, Tom Rymes try...@rymes.com wrote:
On Jan 14, 2011, at 6:45 PM, Bruce B wrote:
You really want to read the LONG LONG signature from some people before
you read the actual latest message? I don't know about thatI guess it's
a preference.
Suffice
It was only the people who ONLY asked in a response to go to Google to find
answers that annoyed me but slowly posting preference adds up as well.
As long as the subject header is not changed all e-mail clients (no matter
how stupid they are), now-a-days, create a nice tree. Even so does
Since I don't want anyone bitch at my spelling again:
news up = nose up :-)
-Bruce
On Fri, Jan 14, 2011 at 8:55 PM, Bruce B bruceb...@gmail.com wrote:
It was only the people who ONLY asked in a response to go to Google to find
answers that annoyed me but slowly posting preference adds up
LOL what a looser. Are you a fat admin behind a desk who is going to loose
his job due to recession and is pissed off?
Here is your first response to one of my first posts:
I was going to respond with some very insightful and helpful information
but I'm not a PRI Guru. Sorry, maybe next time.
, though. I know that
Kamailio would be a very good choice for this role. I believe the
alternatives would be as well.
With kind regards,
Pan B. Christensen
Senior technician
Ibidium AS
http://www.ibidium.no/
- Original Message -
*From:* Bruce B bruceb...@gmail.com
*To:* Asterisk
Yes, it does. Bell provides the same as well and it works with Asterisk.
-Bruce
On Fri, Jan 21, 2011 at 7:11 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Hi list,
For a client I am setting up a system which will use T1 PRI from Primus,
who offer only NI-1 and NI-2 protocols for D-Channels.
Hi Everyone,
I want to call first party using a .callfile and a second party using a
context and then bridge the two calls. I MUST make sure that first party
picks up first and then the second party should be dialed. Trying the
following using an internal extension works nicely and the playback
Hi Everyone,
I don't see any parameter for limiting duration of a call in the .call file
for Asterisk spool outgoing directory.
I'd rather not use a MeetMe to drop the call in a conference room and to
then limit the call duration as that complicates things unnecessarily.
I am wondering if there
Hello,
Are there any other other voice changer applications to Asterisk other than
the one from Lobstertech? (http://lobstertech.com/voice_changer.html)
Specifically interested in open-source but can have a look at economical
commercial alternatives as well.
Thanks
--
AAhem.
https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT
Granted, it's in 1.8, but it's in the documentation ;-)
Cheers
Thanks for the pointer. Unfortunately, I am using 1.6 for all my servers
now. But I would like to know if anyone tested the new pitch changer
On Fri, Jan 28, 2011 at 7:49 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote:
On Friday 28 January 2011 18:27:15 Bruce B wrote:
Hi Everyone,
I don't see any parameter for limiting duration of a call in the .call
file for Asterisk spool outgoing directory.
I'd rather not use a MeetMe
On Mon, Feb 7, 2011 at 8:39 AM, Steve Underwood ste...@coppice.org wrote:
On 02/06/2011 05:05 PM, Sherwood McGowan wrote:
AAhem.
https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT
Granted, it's in 1.8, but it's in the documentation ;-)
Cheers
That seems to do
On Mon, Feb 7, 2011 at 12:40 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
oh and didn't you guys already have your little histrionics sessin about
trimming the goddamned emails, mailing list etiquette about top posting
versus bottom, etc../..
My complaint is not something as
Asterisk runs as root but what about the bash script or the php file that
creates the file? Maybe comment the mv command and check the file
permissions by *ls -la call-filename.call* to be sure.
*chown root.root call-filename* (if root is really the user running
Asterisk) and then the mv command
In my (1.4.X) experience, the file just stays in
/var/spool/asterisk/outgoing and gets “little tags” added until you get the
problem resolved or delete the file.
That is absolutely true if the file is not processed. I guess he can again
do a ls -la in that folder to check permissions for
Thanks Faisal. That is it. I was confused by the fact that there is also the
Context, Extension, and Priority in the .call file that should be filled
along with the Channle: local. I found out that the call file first
calls the local channel context and once that is connected then it moves
Hi everyone,
I know it's off topic from Asterisk directly but yet related.
What sources do you use to limit SIP connecting customers to specific
countries by IP (e.g. allowing USA and not China). It would help me a lot of
you can note the sources you trust that are complete and up to date.
that helps.
Mike.
Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am trying to pass a variable using the .call files but it turns out
blank.
Can someone please point out what might be wrong here:
*/tmp/spool-file.sh
Hi Everyone,
I have a SIP turnk which works fine with both inbound and outbound calling.
However, the only issue is that there is no Ring Tone if someone calls us.
The phones used are Aastra and Polycom connected to the PBX via VPN (SIP).
I do get an outbound ring tone, so it's not that there is
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B
*Sent:* Wednesday, February 16, 2011 2:33 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] No ring tone on inbound call - but
channelconnects fine
Hi Everyone,
I
Hi everyone,
I am reading through Sangoma Wiki right now. But someone may already and
quickly notice this. I have a system that is down since the morning (maybe
power intruptions). All seems fine except for wanrouter status shows
disconnected. Following are the alarms raised. Should I call telco
Hi Everyone,
I have been searching the web and I don't know if SNMP is just that complex
to setup or that not many people use SNMP to monitor Asterisk but the
information is scattered all over. I have got to the point to configure
SNMP with Asterisk and then it's all confusing from there on to
Of *Bruce B
*Sent:* Sunday, March 06, 2011 10:59 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Any good tutorials for setting up Asterisk
SNMP and Cacti for remote monitoring?
Hi Everyone,
I have been searching the web and I don't know if SNMP
Hi everyone,
Installed asterisk from yum repository but I think H.323 is not supported as
I tried commands like this and they don't work:
- *h.323 debug*: Enable chan_h323 debug
- *h.323 gk cycle*: Manually re-register with the Gatekeper
- *h.323 hangup*: Manually try to hang up a call
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