I have tried EyeBeam and it worked fine with x members audio conference
however it need resources (Processing + RAM) per additional line.
Regards,
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Hi Faisal,
Thanks for reply but I want hardware wase VoIP device. If know please
gussed me. From google I fould the list of below devices but I am not sure
that these are best for used or have an issue
*1)Polycom SoundStation IP 7000
*
*Why it's best: *The Polycom SoundStation IP 7000 is
In hardware I used some snom phones up to six lines. You can check on
http://www.snom.com/ for appropriate model.
Regards,
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, November
The Snom 820 handles 5 and that's the highest I've seen in the snom
range.
The snom MeetingPoint (dedicated conference phone) only does 4!
On Wed, 2011-11-30 at 13:55 +0500, Faisal Hanif wrote:
In hardware I used some snom phones up to six lines. You can check on
http://www.snom.com/ for
thank you so much for you help,i have flowed your email and installed
thesesadd-ons all
works perfectly i can store the phone_number of the Customer ,now i can do
what i want :)
thanks every one for your support J
2011/11/30 Dale Noll dn...@wi.rr.com
On 11/28/2011 08:24 AM, salaheddine
Hi everybody,
I' ve been following this list for a while now.
Is there a way to detect the individual and cumulative s/n ratio values for
the incoming calls in Asterisk or any other Call Center solution?...
--
_
-- Bandwidth and
** THIS IS NOT THE RIGHT PLACE TO POST A REPLY **
On Tuesday 29 November 2011, salaheddine elharit wrote:
i use centos 5.5 if i install mysql-devel i can still use the version of
mysql installed now in my server because i use it with a database and Im
afraid to install this mysql-devel and i
I am using my first PAP2 device from linksys. Used many polycom phones...
I configured the PAP2 device with asterisk. I have the registration,
thought I was good to go.
Plugged in my Valcom 2924 public address analog connection, called the
extension
and I got busy... very strange I thought.
On Tue, Nov 29, 2011 at 4:44 PM, john Millican j...@millican.us wrote:
Maybe I am misunderstanding the gist of the comment
OP offered an invalid comparison of how iptables is better than Fail2Ban.
Whether or not OP knew that Fail2Ban simply feeds rules to iptables is
unclear from his comments.
We've been happy with the polycom IP 7000.
Darren Wiebe
On Nov 30, 2011 1:40 AM, virendra bhati virbh...@gmail.com wrote:
Hi Faisal,
Thanks for reply but I want hardware wase VoIP device. If know please
gussed me. From google I fould the list of below devices but I am not sure
that these
On 11/30/2011 09:01 AM, Tom Browning wrote:
I agree - its a bad comparison of 2 different things meant for different
purposes.
iptables is enforcement, fail2ban is detection.
if you have time to sit and make up iptables rules by hand during every
hack attempt
1) you have too much time on
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yasin SULUHAN
Sent: Wednesday, November 30, 2011 6:25 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] s/n ratio detection etc...
Hi everybody,
I' ve been following this
On Wed, Nov 30, 2011 at 4:27 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Yasin SULUHAN
*Sent:* Wednesday, November 30, 2011 6:25 AM
*To:* asterisk-users@lists.digium.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yasin SULUHAN
Sent: Wednesday, November 30, 2011 8:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] s/n ratio detection etc...
On Wed,
Hello,
On one location, I've got from time to time (let say one a week) the
following issue :
the phone SoundPoint 650 works ok (can call or answer, display and sound
are ok),
the sidecar looses its display : entries on sidecar's LCD screen are not
displayed anymore, or names are truncated, or
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, November 30, 2011 9:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Issue with Polycom SPIP650 and its sidecar
Hello,
Hi all,
As a new on asterisk i have some silly questions.
I'll try to connect an asterisk PBX between Telephone provider and an
AVAYA Definity PBX. I've already install elastix-2.2.0 i386 version on
a PC with a DE210 ISDN PRI card.
In previous status we can dial from external directly to
When the side car looses it entries, what does the config file show for the
entries.
This happened to me one time but that was only because for some reason, the
contacts file was deleted by accident and I had to
recreate it. ( I have a backup now too! )
It probably as Dan said, check
2011/11/30 Danny Nicholas da...@debsinc.com
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
*Sent:* Wednesday, November 30, 2011 9:27 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:*
Hi Olivier,
It if occurs only on the sidecar, I would imagine this is either a defective
sidecar/Polycom phone, or a defective PoE switch not giving enough power.
Changing PoE port would eliminate of confirm the PoE port being the issue,
but I'm betting on a Polycom defect.
Make sure the
i have last question regarding this thread
with exten = 3,n,MYSQL(Query resultid ${connid} insert into test (
option_name ) values ('${CALLERID(num)}'))
i can store the phone number without issue
i need also the date and hour fo call in the count coulum
could you please give me the syntex
Out of curiosity, how many concurrent phone calls for an office that uses
Polycoms could be sustained on a DSL ( 3meg down, 768 up )
line using g711?
Not sure if its 64kbps or 87kbps.
I would say roughly 8 but I don't know if the polycoms add any more payload to
the network for presence
Il 30/11/2011 14.38, Jerry Geis ha scritto:
I configured the PAP2 device with asterisk. I have the registration,
thought I was good to go.
Plugged in my Valcom 2924 public address analog connection, called the
extension
and I got busy... very strange I thought
Not so strange ;-)
According to
2011/11/30 eherr email.eherr9...@gmail.com
When the side car looses it entries, what does the config file show for
the entries.
** **
This happened to me one time but that was only because for some reason,
the contacts file was deleted by accident and I had to recreate it. ( I
have a
2011/11/30 Mike l...@net-wall.com
Hi Olivier,
** **
It if occurs only on the sidecar, I would imagine this is either a
defective sidecar/Polycom phone, or a defective PoE switch not giving
enough power. Changing PoE port would eliminate of confirm the PoE port
being the issue, but I’m
Maybe use a power supply instead of PoE, see if problem still occurs. Marco.
Op 30 nov. 2011 18:46 schreef Olivier oza_4...@yahoo.fr het volgende:
2011/11/30 Mike l...@net-wall.com
Hi Olivier,
** **
It if occurs only on the sidecar, I would imagine this is either a
defective
Hello,
the wav sound files that are created by using MixMonitor()-command are
not playable with Windows Media Player.
I can play them with vlc-player and on my Fedora with Totem.
This is one of the files :
/var/ftp/104/2011-11-30_11:54:39_89000404.wav: RIFF (little-endian)
data, WAVE
Since the other data seems kosher, have you tried just renaming the file
without the -, _ and : ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, November 30, 2011 1:55 PM
To: Asterisk Users Mailing List
Hello,
it worked perfectly before... I just did a clean install of my Asterisk
server and changed nothing but Centos 5.6 to CentOS 5.7
Therefore I ask if it should be something that I'm missing on my system ?
Jonas.
On 11/30/2011 08:59 PM, Danny Nicholas wrote:
Since the other data
Check this link - you might be recording a muted file
http://www.centos.org/modules/newbb/print.php?form=1
http://www.centos.org/modules/newbb/print.php?form=1topic_id=34058forum=3
7order=ASCstart=0 topic_id=34058forum=37order=ASCstart=0
From: asterisk-users-boun...@lists.digium.com
Hi All,
How can I find out One way latency from my PBX to my SIP Trunk Provider.
My SIP provider recommends a One way latency of 100ms for good Voice
quality. Ping request to their IP Address gives me a response in approx.
260ms.
Will that be good enough for a SIP Trunk.
Please help. We are
Am 30.11.2011 21:47, schrieb NaJIm:
Hi All,
How can I find out One way latency from my PBX to my SIP Trunk Provider.
My SIP provider recommends a One way latency of 100ms for good Voice
quality. Ping request to their IP Address gives me a response in approx.
260ms.
Will that be good enough
Hi All,
I've been trying to find a solution that would allow our sip phones to
communication with walkie talkies. Our setup is that we have sip phones
setup in 2 locations, headquarters and dome. We can communication from
headquarters and dome through sip phones, but within the dome we have
On Wed, Nov 30, 2011 at 6:20 PM, Ferdinand Babas ba...@cfht.hawaii.edu wrote:
Hi All,
I've been trying to find a solution that would allow our sip phones to
communication with walkie talkies. Our setup is that we have sip phones
setup in 2 locations, headquarters and dome. We can
Am 30.11.2011 21:47, schrieb NaJIm:
Ping request to their IP Address gives me a response in approx. 260ms.
Will that be good enough for a SIP Trunk.
On Wed, 30 Nov 2011, Ruben Rögels wrote:
a ping is the time a packet needs for travelling to a destination and
back to you. So the one way
Thank you Ruben.
Is there anything else that I should be concerned about when looking for a
SIP provider. ??
Regards,
Najim.
On Thu, Dec 1, 2011 at 2:34 AM, Ruben Rögels ruben.roeg...@jumping-frog.org
wrote:
Am 30.11.2011 21:47, schrieb NaJIm:
Hi All,
How can I find out One way
Hi,
I am looking into advising a client on the pro's and cons of using
Installing asterisk on a server vs appliance(e.g digium mypbx). the
appliance seems cheaper initially.
From experience, what would be pro and cons for either option?
--
Best Regards,
James Mutuku Ndeti
Agile Systems
Does that mean I can expect lesser delays with my Voice packets ?? That
would be even better.
Regards,
Najim
On Thu, Dec 1, 2011 at 3:42 AM, Steve Edwards asterisk@sedwards.comwrote:
Am 30.11.2011 21:47, schrieb NaJIm:
Ping request to their IP Address gives me a response in approx.
Is there anything else that I should be concerned about, when looking to
signup for a SIP provider. ??
Regards,
Najim
On Thu, Dec 1, 2011 at 4:49 AM, NaJIm getna...@gmail.com wrote:
Does that mean I can expect lesser delays with my Voice packets ?? That
would be even better.
Regards,
Najim
On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote:
Is there anything else that I should be concerned about, when looking
to signup for a SIP provider. ??
Latency is important, but packet loss also, likewise packet re-ordering.
hw
--
My ping requests show 0% packet loss. How do we find out packet
re-ordering.??
Najim.
On Thu, Dec 1, 2011 at 5:18 AM, Hans Witvliet aster...@a-domani.nl wrote:
On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote:
Is there anything else that I should be concerned about, when looking
to signup
At the most basic level, typically an appliance will have a GUI and be
geared towards non-tech installation. Loading bare Asterisk on a server is
very different. Do you want a GUI or bare Asterisk?
BTW, the MyPBX product is not a Digium product, it's from an oriental
company named Yeastar. My
Hello,
I have written an AGI script for asterisk that uses google translate for
text to speech synthesis.
It supports a variety of different languages, local caching for the voice
data and wideband audio.
The voice in most languages is female and the quality of the synthesized
speech is very high.
a ping is the time a packet needs for travelling to a destination and
back to you. So the one way latency you are refering to, should be half
the time your ping took.
In your case this will be 130ms, I would say this is still reasonable.
I am probably splitting hairs, but that's not always
I've been trying to find a solution that would allow our sip phones to
communication with walkie talkies. Our setup is that we have sip phones
setup in 2 locations, headquarters and dome. We can communication from
headquarters and dome through sip phones, but within the dome we have
On 11/30/2011 11:13 AM, salaheddine elharit wrote:
i have last question regarding this thread
with exten = 3,n,MYSQL(Query resultid ${connid} insert into test (
option_name ) values ('${CALLERID(num)}'))
i can store the phone number without issue
i need also the date and hour fo call in the
WOW.. That is the most complicated Ping I have ever seen.. :)
This is the result I got.
# ping -f -i .02 -s 180 -Q 0xb8 xx.xx.xx.xx
*PING xx.xx.xx.xx (xx.xx.xx.xx) 180(208) bytes of data.
.
--- xx.xx.xx.xx ping statistics ---
15338 packets transmitted, 15325 received, 0% packet loss,
I would bet you get about the same result with the two providers.all
else being equal.
mdev (mean deviation) is a simple way to measure jitter, and you have to
put in context with the min/avg/max numbers. If I had 7ms of deviation
and average times of 4ms, that would be an issue because
Thank you for sharing your exp. with me.
On Wed, Nov 30, 2011 at 7:34 PM, Darren Wiebe dar...@aleph-com.net wrote:
We've been happy with the polycom IP 7000.
Darren Wiebe
On Nov 30, 2011 1:40 AM, virendra bhati virbh...@gmail.com wrote:
Hi Faisal,
Thanks for reply but I want hardware
On 11/30/2011 09:45 PM, Danny Nicholas wrote:
Check this link -- you might be recording a muted file
http://www.centos.org/modules/newbb/print.php?form=1topic_id=34058forum=37order=ASCstart=0
http://www.centos.org/modules/newbb/print.php?form=1topic_id=34058forum=37order=ASCstart=0
Like I
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