Hi Guys,
I am looking to buy a 25 Watt output CyberData VoIP amplifier and to use 2
Bogen sp308a speakers with it for a 40, 000 squar feet area and 21 feet
height. Is that enough? Is there calculator online I can use to determine
the number of speakers needed? I guess these speakers go in chain
Thanks fro the input. The area is a 4 square feet. So, you are saying
that if I use four speakers then they would not be as loud as needed?
Thanks again
2010/7/9 Massimo Nuvoli mass...@archivio.it
bruce bruce ha scritto:
Hi Guys,
I am looking to buy a 25 Watt output CyberData VoIP
The variable is *canreinvite.*
*Please check on voipinfo. If canreinvite is enabled then only SIP signaling
is passed through Asterisk and the media is not passed through Asterisk
resulting in less bandwidth usage and probably less jitter buffer, etcif
you are two phones are closer to each
://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite-Bruce
On Fri, Jul 9, 2010 at 2:40 PM, unsero...@aol.com wrote:
Sounds great, thanks for your answer.
Do i need to set this on the trunk, the friend or on both?
-Original Message-
From: bruce bruce bruceb...@gmail.com
To: Asterisk
Hi Guys,
I am making another module for Voicemail. I have three fields in a POST form
that have to be connected together to make it a single 10 digit number but
there is something wrong in my syntax probably.
$npaa = ('$_POST[anpa]');
$nxxa = ('$_POST[anxx]');
$blocka = ('$_POST[ablock]');
Hi Everyone,
I am trying to find the issue of dropped calls in the middle of the
conversation. The system is Elastix. Anyway to know which party hangup the
channel in case of Asterisk 1.4 and Sangoma analogue cards? (this is not
PRI)
Thanks,
Bruce
--
[mysqld_safe]
log-error=/var/log/mysqld.log
pid-file=/var/run/mysqld/mysqld.pid
log=/var/log/mysql_query.log
*But it doesn't log anything to /var/log/mysql_query.log*
What else am i missing?
Thanks
On Thu, Jul 8, 2010 at 1:20 AM, Steve Edwards asterisk@sedwards.comwrote:
On Thu, 8 Jul 2010, bruce
Hi Everyone,
I want to fine tune the Rx and Tx gain on an analogue Sangoma card by
dialing into another server that is running on Sangoma PRI card (both
services on Bell network).
[mwatt1004khz]
exten = s,1,Answer
exten = s,n,PlayTones(1004/1000)
exten = s,n,Wait(300)
If I match the Rx/Tx
Jul 2010, bruce bruce wrote:
I have this in /etc/my.cnf:
[mysqld]
datadir=/var/lib/mysql
socket=/var/lib/mysql/mysql.sock
user=mysql
old_passwords=1
log-error=/var/log/mysqld.log
[mysqld_safe]
log-error=/var/log/mysqld.log
pid-file=/var/run/mysqld/mysqld.pid
log=/var/log
Thanks for the input guys. My client is looking for Y-cords to train people.
So, set beside them take a call and let them listen on the other call. They
currently use wireless Plantronic headset with Aastra phones. Can you
suggest any specific vendors for Y-cords?
Thanks
On Tue, Jul 6, 2010 at
Hi Guys,
This is something related and yet un-related to Asterisk. I have a
FreePBX/Asterisk server running and I want to trace everything that FreePBX
does to MySQL. Is there a verbose CLI to MySQL that I can pull up on
terminal and make configuration change to FreePBX and see it in real-time on
the name of the mysql log file is.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-07 9:43 PM, Carlos Chavez cur...@telecomabmex.com wrote:
*On Wed, 7 Jul 2010 19:19:28 -0400, bruce bruce wrote*
Hi Guys,
This is something related and yet un-related to Asterisk. I have
Just downloaded PrivateSHELL and it seems to be what everyone is looking for
in Putty. It's much better than putty in terms of not being sluggish and
scrolling is fine. Plus the window and the text doesn't hurt your eyes. It
has One click SFTP as well. So, good bye to WinSCP.
I think I found what
Good Afternoon,
Can someone please explain what Y-cords are available out there and how they
can be used with Aastra or other VoIP phones? Maybe with or WITHOUT
headsets?
Isn't a Y-cord traded for soft Barge in these days?
Thanks,
Bruce
--
Hi Guys,
I have a channel that is dialed with *Timeout* option. So, there is definite
time to it. Only thing is that I don't have control of that channel. I only
know that it's using g729 codec and that there is only one channel that is
using g729 at any given time. So, my question is:
From
Anything guys?
Thanks
On Mon, Jun 28, 2010 at 10:20 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
I want to know a bit about the guts of the current AsterisNOW system. I
know that FreePBX is embraced as the main GUI but is just an install of
CentOS 5.4 + (Asterisk/FreePBX from
Hi guys,
I have two Asterisk servers (with FreePBX) connected together with IAX2
trunking. When I call from server A-B call connects but hangs up after 30
seconds. What could be cause?
Can anyone please share working configuration between two asterisk server in
IAX2 trunking for FreePBX?
Thanks
remaining EPOCH time?
Thanks
On Sun, Jul 4, 2010 at 12:03 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sun, 4 Jul 2010, bruce bruce wrote:
I have a channel that is dialed with Timeout option. So, there is definite
time to it. Only thing is that I don't have control of that channel. I
06:53 AM, bruce bruce wrote:
Hi Everyone,
I am accustomed to PUTTY and it's very nice as in it allows many many
SSH profiles to be saved and allows tunneling etcbut it's not very
good when it comes to scrolling up and down, colors, text size, and
specially it doesn't give a title
Thanks a lot. I will look into it.
On Wed, Jun 30, 2010 at 11:15 AM, Warren Selby wcse...@selbytech.comwrote:
On Wed, Jun 30, 2010 at 8:40 AM, bruce bruce bruceb...@gmail.com wrote:
Thanks a lot.
-Bruce
On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone carbe...@gmail.comwrote:
Hi bruce
Yes, you are missing a whole bunch of configurations from creating SIP users
to making sure they show as peers on Asterisk to making sure you use dnid,
etc.You probably might want to search google for some configuration help
On Wed, Jun 30, 2010 at 11:24 AM, gokulakrishnan
1 name for authentication with proxy server
line1_authname : 246
; Line 1 authentication name password
line1_password : afjhajshdga
; Phone Label (Text desired to be displayed in upper right corner)
phone_label: XX246
i hope this help you!
regards
2010/6/30 bruce bruce bruceb
Hi Everyone,
I am accustomed to PUTTY and it's very nice as in it allows many many SSH
profiles to be saved and allows tunneling etcbut it's not very good when
it comes to scrolling up and down, colors, text size, and specially it
doesn't give a title to the opened instance. Maybe giving the
I have an *ipphone 7965G* which has to be connected to Asterisk. It has been
flashed with SIP firmware but the config file doesn't seem to work maybe I
am missing something in it.
I appreciate it if you can share your working sample config file with me.
Thanks
--
Hi Everyone,
I want to know a bit about the guts of the current AsterisNOW system. I know
that FreePBX is embraced as the main GUI but is just an install of CentOS
5.4 + (Asterisk/FreePBX from Yum repos)?
- Or is there anymore to this? Maybe some security tools?
- Or is Asterisk built from the
Hi Guys,
Asterisk 1.6.2.7 install from Yum Repository shows a lot of : doing
dnsmgr_lookup for sip.provider.com
Google searches show it was fixed in some version.
Is this to be ignored?
Thanks
--
_
-- Bandwidth and
It's one of the bad modules that goes with FreePBX anyhow. The moment you go
over 3000 recordings you are already in trouble. It's about time someone
come up with a better moduel.
On Wed, Jun 23, 2010 at 11:05 AM, Mickael Monsieur
mickael.monsi...@gmail.com wrote:
Hello,
I look ARI (Asterisk
://www.xorcom.com/downloads/astribank2-trixbox-ce-drivers.html#trixboxce2.8
On Tue, Jun 22, 2010 at 5:25 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Jun 22, 2010 at 12:58:00AM -0400, bruce bruce wrote:
Hi Guys,
An 8 channel
FXO?
Astribank is connected to Trixbox 2.8 and I
that the fact that it is so slow exposes its raciness[1].
On 21 June 2010 16:08, bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
I want to know if a specific codec type is used at least one. For
example,
I want to know if out of the 100 calls on the system if there is a 1
channel
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce
*Sent:* Tuesday, June 22, 2010 1:32 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x
Hi Everyone,
I was on Xorocom site but there is no clear and consice place to download
drivers and firmware. I am reading their instructions to install Astribank 8
channel FXO on Trixbox 2.8 and I seem to be missing files at this step:
[pbx.archology.com dahdi]# /usr/share/doc/astribank_upgrade
Trixbox
from Elastix and version to version.
On Tue, Jun 22, 2010 at 4:53 PM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 22 Jun 2010, bruce bruce wrote:
I was on Xorocom site but there is no clear and consice place to
download drivers and firmware. I am reading their instructions
Hi Everyone,
I want to know if a specific codec type is used at least one. For example, I
want to know if out of the 100 calls on the system if there is a 1 channel
that is running G.729 codec right now. If using dial-plan and I dial in, I
can use this to obtain information about CURRENT channel.
Hi Guys,
An 8 channel Astribank is connected to Trixbox 2.8 and I ran
freepbx-module-zapauto but I get the following when running these
commands and can't make calls out:
[Trixbox]# dahdi_genconf xpporder
/usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing
Hi Guys,
I am looking to delete some of the CDR logged by Asterisk in asteriskcdrdb
in a PbxinaFlash system running Asterisk 1.4.x
The CDR records to deleted are probably a big chunk and spread out all
through the database but I basically want to delete all calls that came in
through a specific
Hi Guys,
Is it possible to harvest the output of system into a SetVar(variable)?
exten = s,n,SetVar(var=system(*asterisk -rx sip show channels | grep -c
(ulaw))*
*
*
*??? any problem with the syntax? *
*
*
*
*
*Thanks,*
*
*
--
:58:17AM -0400, bruce bruce wrote:
Hi Guys,
Is it possible to harvest the output of system into a SetVar(variable)?
exten = s,n,SetVar(var=system(*asterisk -rx sip show channels | grep
-c
(ulaw))*
There's the function SHELL. Though I suspect you use 1.2 and I'm not
sure
Nice and colorful tutorial for cronjobs.
http://www.linuxconfig.org/Linux_Cron_Guide
-Bruce
On Fri, Jun 18, 2010 at 1:55 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
thanks for your response
how can i create and execute this cron
2010/6/18 Danny Nicholas da...@debsinc.com
Hi Everyone,
I have a php file that if an argument is passed to it, it will echo a number
back. I am looking to use system() in dial-plan to send ${EXTEN} to it and
then to get that processed value back from the php file and put it in $var
back into asterisk dial-plan. While trying this method
, Jun 14, 2010 at 12:12 PM, Carlos Chavez cur...@telecomabmex.comwrote:
On Mon, 2010-06-14 at 12:00 -0400, bruce bruce wrote:
Hi Everyone,
I have a php file that if an argument is passed to it, it will echo a
number back. I am looking to use system() in dial-plan to send
${EXTEN
-0400, bruce bruce wrote:
Hi Carlso,
Thanks for the input. I have done this in php and am not familiar with
phpagi.
So, there is absolutely no way to temporarily solve this problem by
getting the value back from php file?
Wondering if it would require a lot of work to change
to dialplan
exit(0);
?
Thanks,
Bruce
On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez cur...@telecomabmex.comwrote:
On Mon, 2010-06-14 at 13:41 -0400, bruce bruce wrote:
Hi Carlso,
Thanks for the input. I have done this in php and am not familiar with
phpagi.
So, there is absolutely
sent to the AGI script from Asterisk. I do not know why
you are getting the channel instead of the extension, you could try
giving the extension as a parameter to the AGI script if you cannot get
that from the included request variable.
On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote
Hi Guys,
Looking for a powerful box that is compact, can take two hard drives for
Raid-1 (no SSD, too expensive), have at least two Gig ports or two
10/100mbps ports. Fit two PCIe or one PCIe card plus it's daughter card
which needs as much room as a PCIe and doesn't need the actual slot. That is
notice the last three charecters
of that line is* )*. So, when the phpagi path is correct, it looks like:
*415444555)*.
-Bruce
On Mon, Jun 14, 2010 at 6:09 PM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 2010-06-14 at 14:57 -0400, bruce bruce wrote:
Carlos, Thanks a lot for getting me
-rx sip show peer $sip_peer | grep -c
X-Lite', $retval);
Should $sip_peer be inside another set of parenthesis?
Thanks,
Bruce
*
On Mon, Jun 14, 2010 at 6:44 PM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 14 Jun 2010, bruce bruce wrote:
Thanks for the input. I actually did use
Hi Guys,
I have Spikko setup as provider of DID and outbound routes and I can make
calls out but no inbound calls via DID can be made. I did a sip debug which
is reported below. I never receive the call though, I have a catch all in my
inbound routes and it doesn't hit my context at all or not
Since you mentioned FreePBX, unfortunately, it's not only the GUI that
drives the system and it can be that at some point someone planted
the extension in one of your .conf or other file if they had access to SSH
or some other way.
Going back to occurrence in sip.conf as mentioned, of course
# if test x$CONSOLE != xno ; then
# ASTARGS=${ASTARGS} -c
# fi
#fi
On Mon, Jun 7, 2010 at 8:12 PM, bruce bruce bruceb...@gmail.com wrote:
I did see the TTY=9 on the third or fourth line but commenting that doesn't
help much. I would really appreciate it if you can send
? Are
there any errors in the asterisk logs? Does asterisk stay running after it
starts?
~Seann
On 6/6/2010 5:00 PM, bruce bruce wrote:
Reboot like 10 times and the problem still presists.
Also, upon reboot despite having done chkconfig --add asterisk asterisk
still doesn't load automatically
macro-vm, extension vmx!*
Thanks,
Bruce
On Mon, Jun 7, 2010 at 3:29 PM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 7 Jun 2010, bruce bruce wrote:
CentOS 5.4 and asterisk does stay running after it's loaded by asterisk
-g. But the chkconfig --add asterisk doesn't work :(
What does
I did see the TTY=9 on the third or fourth line but commenting that doesn't
help much. I would really appreciate it if you can send the changes you
made.
Indeed it is a VPS.
Thanks,
Bruce
On Mon, Jun 7, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote:
*chown: cannot access
Hi Guys,
Just did an Asterisk 1.6.x (repo install) and FreePBX (source install). When
trying to dial a number, I get this:
tel*CLI Use of uninitialized value in hash element at /var/www/html/panel/
op_server.pl line 3367.
Use of uninitialized value in concatenation (.) or string at
Hi Guys,
I have tried every single rule I could into iptables but I can't register
this VPS to a provider Spikko. Finally I did an iptable accept on INPUT,
OUTPUT, and FORWARD, for ports 0:65000 just to test things and still I can't
register to the provider.
I can easily register to another
for the input.
On Sun, Jun 6, 2010 at 4:47 PM, dotnetdub dotnet...@gmail.com wrote:
On 6 June 2010 19:48, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
Just did an Asterisk 1.6.x (repo install) and FreePBX (source install).
When trying to dial a number, I get this:
tel*CLI Use of uninitialized
, Tilghman Lesher tles...@digium.com wrote:
On Sunday 06 June 2010 13:46:49 bruce bruce wrote:
I have tried every single rule I could into iptables but I can't register
this VPS to a provider Spikko. Finally I did an iptable accept on INPUT,
OUTPUT, and FORWARD, for ports 0:65000 just to test
know, in 1.6 is no more call-limit in sip.conf
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
bruce bruce wrote
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
bruce bruce wrote:
Thanks for the advice, but I have to keep the customer on hold till the
line becomes available
?
Thanks
On Sat, May 29, 2010 at 7:07 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sat, 29 May 2010, bruce bruce wrote:
I am looking to use System() function along with some bash scripting to
determine if a Trunk is being used during certain time of the day or
not. Here is what I have
, May 30, 2010 at 9:37 AM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the tip. I have been checking those two options. Would you be
able to provide an example of how GROUP or GROUP_COUNT may check for a
trunk
usuage?
Here is how I do it. It is based on Asterisk 1.6.1.x, and I created
Hi Guys,
Anyone else can comment on this or give me their thoughts please? I just
want to know if someone can confirm the output for make install in new
LibPRI directory.
Thanks,
Bruce
On Fri, May 28, 2010 at 12:58 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the input. Yes, I did
Hi Guys,
I am looking to use System() function along with some bash scripting to
determine if a Trunk is being used during certain time of the day or not.
Here is what I have in mind. Please guide me if you know a better way:
exten = s,1,answer
exten = s,n,System(/tmp/check.sh)
check.sh:
check
be really helpful.
Thanks,
Bruce
On Sat, May 29, 2010 at 5:28 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Should be solid. After all munin also works on the same lines and it works
solid.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-05-29 5:12 PM, bruce bruce
:
- bruce bruce bruceb...@gmail.com wrote:
What am I doing wrong that it's not update to 1.4.11?
Thanks, Bruce
--
Did you restart your services to ensure the new library was picked up?
--Tim
--
_
-- Bandwidth and Colocation
Hi Guys,
I am running a PBXinaFLASH server. I replaced contents of /usr/src/libpri
with the new version of Libpri v1.4.11. The installed one was v1.4.10.
System is running Asterisk 1.4.21.2.
I did the following after:
cd /usr/src/libpri/
make
make clean
make install
Install end with these
Thanks for the update. How to upgrade to the latest stable release without
compliling Asterisk again? Can you please explain and detail the commands?
We are running PBXinaFlash with LibPRI 1.4.10.1 which gives lots of
problems.
Thanks
On Wed, May 26, 2010 at 12:27 PM, Asterisk Development Team
to
strip or hide the CLID if Callee requested private presentation?
Thanks
On Sat, May 15, 2010 at 4:14 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
We have a problem with Caller ID not being displayed. I want to test
everything to see where the problem is with the incoming Caller ID
Is the Java soft phone an open source or obtainable? I am just checking
their site and it seems they only provide service??!!
Their java web based client is built neatly. Would like to test that on my
servers.
On Thu, May 20, 2010 at 3:21 PM, mgra...@mstvp.com wrote:
I've used HP Thin Clients
That is the RFC number for SIP. Yes, Asterisk is compliant with RFC. I am
not sure to what degree but I haven't ever faced non-compliance on SIP RFC 3261
ever with any provider.
-Bruce
On Wed, May 19, 2010 at 2:28 PM, Tarek Sawah tareksa...@hotmail.com wrote:
Greetings List,Trying to
Hi Guys,
Running the following with a Sangoma A101D PRI card:
*Asterisk 1.4.21.2*
*LibPRI version: 1.4.10*
No inbound or outbound calls can be made. In fact Asterisk CLI doesn't show
any activity. Problem goes away on restart of the system or maybe asterisk.
I see post about upgrading Libpri to
Hi Guys,
I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in the
current 1.4.10 version. I am running Asterisk 1.4.x (in fact it is a
PBXinaFLASH system).
How can I upgrade to the latest Libpri? Do I need to re-install Asterisk?
Won't that break the box?
Can I simply do this
17, 2010 at 3:48 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, May 17, 2010 at 03:22:04PM -0400, bruce bruce wrote:
Hi Guys,
I have to upgrade to latest Libpri 1.4.10.2 due to an existing bug in the
current 1.4.10 version. I am running Asterisk 1.4.x (in fact
/lib ; ln -sf libpri.so.1.4 libpri.so)
install -m 644 libpri.a /usr/lib
if test $(id -u) = 0; then /sbin/ldconfig -n /usr/lib; fi
Thanks,
Bruce
On Mon, May 17, 2010 at 4:03 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the help Tzafrir.
I think for libpri you meant = 1.4.x rather than
, May 15, 2010 at 4:56 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Sat, May 15, 2010 at 04:32:19PM -0400, bruce bruce wrote:
Hi Guys,
Can q931.c be re-compiled using gcc or something else without the need to
re-do the whole libpri? Some changes were made to q931.c and I want those
Maybe drop the call in a Meetme room and have an announcement?
On Sun, May 16, 2010 at 10:15 AM, Bruce Ferrell bferr...@baywinds.orgwrote:
I'm trying to make an AMI call. I want to call a number, play an
announcement when the call is answered, then call a second number and
connect the two
Hi Guys,
We have a problem with Caller ID not being displayed. I want to test
everything to see where the problem is with the incoming Caller ID and why
it's not displaying.
I am tracking this down to Presentation prohibited of network provided
number even though the Caller doesn't use *67 and
Hi Guys,
Can q931.c be re-compiled using gcc or something else without the need to
re-do the whole libpri? Some changes were made to q931.c and I want those to
be reflected in .a .o .so .lo files as I think those are the files read by
Asterisk vs the .c file.
Thanks,
--
Hello Everyone,
Are these indications of attacks on this system? I specifically have port 22
disabled at all times and only port forward it to server when I access SSH
for a minute or so. Shouldn't UNKNOWN be an actual IP address?
*/var/log/secure:*
May 14 00:35:39 pbx sshd[9011]: Did not
Unplugging just turns off the phone and has no effect on the settings. You
can not damage the phone by tampering configurations but you can mess up
the settings and it might not register, send, or receive calls.
User manu for your reference:
Hi Guys,
Anyone might know why this error keeps showing up and inbound/outbound is
not working on a Bell PRI with Sangoma A101D?
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement
No calls can be made inbound/outbound.
Keeps repeating. No alarms ON and no changes been made to
with the
other party.
You are sending FACILITY - maybe the other party does not like FACILITY and
hangs up.
IIRC there is a setting in zapata.conf to enable/disable FACILITY.
regards
klaus
Am 10.04.2010 21:46, schrieb bruce bruce:
Hi Guys,
I am calling out 416-999- on Channel 1 of PRI
Hi Everyone,
How is this possible? How can I go about debugging this? I think that the
sound chopping and choking is also related to this. I have never seen
Asterisk show 43% of cpu usuage when there is only one call going. It
actually flactuates down to 11% and up to 43%.
Please guide me as to
Adobe Air and Adobe FMS are good examples of VoIP working flawlessly over
TCP. We are actually developing a flash phone which needs only TCP to
transmit both signal and audio.
-Bruce
On Sat, Apr 24, 2010 at 2:01 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
RTP stands for Real-Time Transport
WWW-Authenticate: Digest algorithm=MD5, realm=103001vc, nonce=03e68412
Content-Length: 0
Jonas.
bruce bruce wrote:
Try changing port=5064 to port=5060 in your Asterisk config file. Portech
will negotiate it's port with Asterisk itself
Take out the router/firewall and connect directly to the net to test your
NAT problem theory.
On Thu, Apr 22, 2010 at 12:15 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
Jared,
thank you for your answer.
As I said in my previous mail, I'm using a Zyxel NBG-419 router (which
normally
I have a list of CLIDs prefixes that I want to use in a context.
Basically, I want to do this but the list of prefix numbers is much longer.
List of prefixes (556,557,557,989.)
[custom-inbound]
exten = _556,1,answer
exten = _556,n,playback(beep)
exten = _557,1,answer
exten =
Hi Everyone,
I have a weired situation where calls in and out are proceessed all right
but when I dial *97 Asterisk is literally choking when it comes to
announcements like Password or Call from 205-456-. Each one of those
announcements can take like 10+ seconds to finish with most of it not
Yes, it's all g.711 ulaw.
On Wed, Apr 21, 2010 at 1:37 PM, Darrick Hartman (lists)
dhart...@djhsolutions.com wrote:
Are your sound files being transcoded or played back in their native
formats?
On 04/21/2010 12:25 PM, bruce bruce wrote:
Hi Everyone,
I have a weired situation where
yes, it's on Amazon.
On Wed, Apr 21, 2010 at 2:26 PM, Ryan Bullock rrb3...@gmail.com wrote:
Are you running asterisk in a virtual machine?
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Thanks for the input.
I am going to check this once I get access to system again tonight.
But I thought the timing source dahdi_dummy is only good for features like
MeetMe or conference rooms? or am I wrong and it has an effect on any type
of calls and checking voice messages?
Thanks
On Wed,
tell from these?
On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce bruceb...@gmail.com wrote:
Thanks for the input.
I am going to check this once I get access to system again tonight.
But I thought the timing source dahdi_dummy is only good for features like
MeetMe or conference rooms? or am I
at 7:56 PM, Sean Brady sbr...@gtfservices.com wrote:
On 04/21/2010 05:36 PM, bruce bruce wrote:
Here are result of dahdi_test:
[r...@ip-10-251-123-3 ~]# dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434%
-434.763
I know that anything lower than 99% is bad. But *-400 *?
Anything care of comment?
Thanks,
On Wed, Apr 21, 2010 at 7:45 PM, Steve Howes steve-li...@geekinter.netwrote:
On 22 Apr 2010, at 00:36, bruce bruce wrote:
Opened pseudo dahdi interface, measuring accuracy...
99.725% 96.018% 99.532
at anytime on this server.
Thanks
On Wed, Apr 21, 2010 at 8:03 PM, Carlos Chavez cur...@telecomabmex.comwrote:
On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote:
Here are result of dahdi_test:
[r...@ip-10-251-123-3 ~]# dahdi_test
Opened pseudo dahdi interface, measuring accuracy
I have had problems with Portech firmware using Chrome browser. The problem
was that when I changed the password on the gateway it would apply that
password to SIP PEERS as well. So, maybe, you are actually not having the
right password in your SIP peer as well and hence your Asterisk sends
a charm with an IP-phone
(Grandstream) ?!
Jonas.
bruce bruce wrote:
I have had problems with Portech firmware using Chrome browser. The problem
was that when I changed the password on the gateway it would apply that
password to SIP PEERS as well. So, maybe, you are actually not having
, Apr 19, 2010 at 12:08 AM, Alyed al...@vivoxie.com wrote:
You can't do that with Xlite, try Sjphone instead.
Alyed
2010/4/17 bruce bruce bruceb...@gmail.com
Hi Guys,
Wondering if anyone has tried to make a direct SIP peer to peer call using
x-lite without any registrations of any sort. I
Hello Everyone,
I have a system that was working on Sunday 1 P.M. and then gives Congestion
on Monday morning. Sometimes over night it probably stopped working. It's a
PBXinaFLASH with Asterisk 1.4 and libPRI with a 23 channel PRI connected and
24th D-Channel.
This is all I see in
I've never been able to with xlite
it's just with Sjphone it's straight forward.
Alyed
2010/4/19 bruce bruce bruceb...@gmail.com
That is not correct. It's possible by adding a display name and adding the
IP address of the pbx you are calling as the host ip. Then uncheck the
register button
Dial: 0
Logical Channel Mapping: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T309 Timer: -1
T313 Timer: 4000
N200 Counter: 3
thanks,
Bruce
On Mon, Apr 19, 2010 at 3:00 PM, Doug Lytle supp...@drdos.info wrote:
bruce bruce wrote:
[2010-04-19 08:45:50] WARNING
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