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]')";
<http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite>-Bruce
On Fri, Jul 9, 2010 at 2:40 PM, 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
>
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 othe
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
> bruce bruce ha scritto:
> > Hi Guys,
> >
> > I am looking to buy a 25 Watt output CyberDa
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 so
Putting it in /tmp/ just did the job. Sorry, I posted my older my.cnf file.
I actaully did have the log under mysqld rather than the safe version but it
didn't work. I will put this to privilage problems.
On Thu, Jul 8, 2010 at 9:55 PM, Steve Edwards wrote:
> On Thu, 8 Jul 2010, bru
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 num
[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 wrote:
> On Thu, 8 Jul 2010, bruce bru
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
--
tever the name of the mysql log file is.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-07-07 9:43 PM, "Carlos Chavez" wrote:
>
> *On Wed, 7 Jul 2010 19:19:28 -0400, bruce bruce wrote*
>
> > Hi Guys,
> >
> > This is someth
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
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 4:
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
--
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
or MS
> Windows platform and WHY?
>
> On 06/29/2010 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
> &g
I find out it's remaining EPOCH time?
Thanks
On Sun, Jul 4, 2010 at 12:03 PM, Steve Edwards wrote:
> 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 cont
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
Anything guys?
Thanks
On Mon, Jun 28, 2010 at 10:20 PM, bruce bruce 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
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 with
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 wrote:
> Hi All,
>
Thanks a lot. I will look into it.
On Wed, Jun 30, 2010 at 11:15 AM, Warren Selby wrote:
> On Wed, Jun 30, 2010 at 8:40 AM, bruce bruce wrote:
>
>> Thanks a lot.
>>
>> -Bruce
>>
>>
>> On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone wrote:
>>
&
irectory_url: "http://192.168.20.4/xmlservices/phonebook.xml";
>
> logo_url: "http://192.168.20.4/images/logo.bmp";
>
> SIP_MAC_ADDR.conf
>
> proxy1_address: 192.168.20.4
>
> ; Line 1 phone number
> line1_name : 246
>
> ; Line 1 name for authent
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 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
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 so
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 Coloca
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 (Asteri
arate Trixbox
from Elastix and version to version.
On Tue, Jun 22, 2010 at 4:53 PM, Steve Edwards wrote:
> 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 inst
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
/
and rewrite it in C. thoughts>
>
>
>
> -Elliot
>
>
>
> *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-
gt; > native;
> > done
>
> The overhead of each 'asterisk -rx' command is noticable. If you have 10
> calls or more, this can have an odd effect.
>
> Not to mention that the fact that it is so slow exposes its raciness[1].
>
> >
> > On 21 June 2010 16:08,
ttp://www.xorcom.com/downloads/astribank2-trixbox-ce-drivers.html#trixboxce2.8>
On Tue, Jun 22, 2010 at 5:25 AM, Tzafrir Cohen wrote:
> On Tue, Jun 22, 2010 at 12:58:00AM -0400, bruce bruce wrote:
> > Hi Guys,
> >
> > An 8 channel
>
> FXO?
>
> > Astri
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
'/sys/bus/xpds/devices/00:0:0/timing_
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,
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 D
with first three digits being 789 (for example) as all of those
variables can help me move to next step which is to decide to place a second
call through the same trunk or not.
Any inputs?
Thanks a lot
On Sat, Jun 19, 2010 at 1:56 PM, Tzafrir Cohen wrote:
> On Sat, Jun 19, 2010 at 10:58:17AM
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,*
*
*
--
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
>
>> I do a c
is always returned
as 0.
$peer_count = system('asterisk -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 wrote:
> On Mon, 14 Jun 2010,
right because if you 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 wrote:
> > 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
hat are
> automatically 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, 20
n Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez wrote:
> 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 absol
EN})
*415444555*
But with the agi_extension it comes back as:
NoOp("SIP/64.111.222.111-0ca7", "")
Where can I find the list of command requests that can be sent to Asterisk?
Specially that for DID.
Thanks
On Mon, Jun 14, 2010 at 2:15 PM, Carlos Chavez wrote:
> O
, Jun 14, 2010 at 12:12 PM, Carlos Chavez wrote:
> 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
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 $var
back into asterisk dial-plan. While trying this method doe
Hi Guys,
I have Spikko setup as provider of DID and outbound routes and I can make
calls out but no inbound calls via DID can be made. I did a sip debug which
is reported below. I never receive the call though, I have a catch all in my
inbound routes and it doesn't hit my context at all or not sip
fi
# ASTARGS="${ASTARGS} -vvvg"
# if test "x$CONSOLE" != "xno" ; then
# ASTARGS="${ASTARGS} -c"
# fi
#fi
On Mon, Jun 7, 2010 at 8:12 PM, bruce bruce wrote:
> I did see the TTY=9 on the third or fourth line but com
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
FreeP
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 wrote:
>
> *chown: cannot access `/dev/tty9': No such file or
emove this priority label
(57/vmxopts) from the peer_label_table of context macro-vm, extension vmx!*
Thanks,
Bruce
On Mon, Jun 7, 2010 at 3:29 PM, Steve Edwards wrote:
> On Mon, 7 Jun 2010, bruce bruce wrote:
>
> > CentOS 5.4 and asterisk does stay running after it's loaded b
re you using? 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 don
58 PM, Tilghman Lesher 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
t be
related?
Thanks for the input.
On Sun, Jun 6, 2010 at 4:47 PM, dotnetdub wrote:
>
>
> On 6 June 2010 19:48, bruce bruce wrote:
>
>> Hi Guys,
>>
>> Just did an Asterisk 1.6.x (repo install) and FreePBX (source install).
>> When trying to dial a number,
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 provi
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
/var/www/html
an 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:
> > Thanks for the
web.
>
> And how I 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: + 37
On Sun, May 30, 2010 at 1:47 PM, Jonathan Thurman
wrote:
> On Sun, May 30, 2010 at 9:37 AM, bruce bruce 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
> >
?
Thanks
On Sat, May 29, 2010 at 7:07 PM, Steve Edwards wrote:
> 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 wha
A bit detail would be really helpful.
Thanks,
Bruce
On Sat, May 29, 2010 at 5:28 PM, Zeeshan Zakaria 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.
>
&
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
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 wrote:
> Thanks for the input. Yes, I did do
Thanks for the input. Yes, I did do a restart of Asterisk and the system but
changes do no show. Is it normal to not see a "Successful" message after
doing "make install" within the new LibPri library?
Thanks,
Bruce
On Thu, May 27, 2010 at 9:41 PM, Tim Nelson wrote:
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 line
strip or hide the CLID if Callee requested private presentation?
Thanks
On Sat, May 15, 2010 at 4:14 PM, bruce bruce 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 and wh
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 <
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, wrote:
> I've used HP Thin Clients as embedded hosts
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 wrote:
>
>
> Greetings List,Trying to interconnect with a new pro
| grep -q "enabled"); then /sbin/restorecon -v
/usr/lib/libpri.so.1.4; fi
( cd /usr/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 wrote:
>
Mon, May 17, 2010 at 3:48 PM, Tzafrir Cohen wrote:
> 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
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 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
Maybe drop the call in a Meetme room and have an announcement?
On Sun, May 16, 2010 at 10:15 AM, Bruce Ferrell wrote:
> 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 when the second call
nks again,
Bruce
On Sat, May 15, 2010 at 4:56 PM, Tzafrir Cohen wrote:
> 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 w
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,
--
_
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
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:
http://www.aastra.com/cps/rde/xbcr/SID-3D8CCB6A-E9BBA
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 receiv
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 th
a problem 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 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 w
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 wrote:
> RTP stands for Real-Time Transport Protocol. TCP is no
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 => _557,n,playb
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 wrote:
> Jared,
>
> thank you for your answer.
>
> As I said in my previous mail, I'm using a Zyxel NBG-419 router (which
> normally supports VoIP and QoS
IFY
> Supported: replaces
> WWW-Authenticate: Digest algorithm=MD5, realm="103001vc", nonce="03e68412"
> Content-Length: 0
>
>
> Jonas.
>
> bruce bruce wrote:
>
>
ad at anytime on this server.
Thanks
On Wed, Apr 21, 2010 at 8:03 PM, Carlos Chavez wrote:
> 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, me
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 wrote:
> On 22 Apr 2010, at 00:36, bruce bruce wrote:
> > Opened pseudo dahdi interface, measuring accuracy...
> > 99.725% 96.018% 99.532%
d, Apr 21, 2010 at 7:56 PM, Sean Brady 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%
tell from these?
On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce 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
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, Ap
yes, it's on Amazon.
On Wed, Apr 21, 2010 at 2:26 PM, Ryan Bullock wrote:
> Are you running asterisk in a virtual machine?
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join
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 Ever
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 n
al registration succeeds and to a public server fails...
> NAT anyone ? But then why does it work like a charm with an IP-phone
> (Grandstream) ?!
>
>
> Jonas.
>
>
>
> bruce bruce wrote:
>
> I have had problems with Portech firmware using Chrome browser. The problem
> was t
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
Unautho
1
Retrans: 0
Busy: 0
Overlap 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 wrote:
> bruce bruce wrote:
> >
> >
ot;I've never been able to with xlite"
> it's just with Sjphone it's straight forward.
>
> Alyed
>
> 2010/4/19 bruce bruce
>
> That is not correct. It's possible by adding a display name and adding the
>> IP address of the pbx you are calling as th
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 /var/log/asteri
-Bruce
On Mon, Apr 19, 2010 at 12:08 AM, Alyed wrote:
> You can't do that with Xlite, try Sjphone instead.
>
> Alyed
>
>
> 2010/4/17 bruce bruce
>
>> Hi Guys,
>>
>> Wondering if anyone has tried to make a direct SIP peer to peer call using
>> x
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