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
That looks like the option that will help a lot.
Thanks.
On 8 July 2010 23:21, Steve Edwards wrote:
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
>> Lyndon-Smith
>>
>> We have had 20 calls over the last month where the SIP channel has not
>> identified that the person o
Zeeshan Zakaria ha scritto:
> I have two test asterisk boxes, both version 1.4.26, on which I do
> Answer() followed by MusicOnHold() and it works just fine. I do this all
> the time as this is my standard way of testing new contexts.
Yesterday i tested another installation and i found the same is
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, bruce bruce wrote
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
On Thu, 8 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
> l
I have this in /etc/my.cnf:
[mysqld]
datadir=/var/lib/mysql
socket=/var/lib/mysql/mysql.sock
user=mysql
# Default to using old password format for compatibility with mysql 3.x
# clients (those using the mysqlclient10 compatibility package).
old_passwords=1
log-error=/var/log/mysqld.log
[mysqld_
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
> Lyndon-Smith
>
> We have had 20 calls over the last month where the SIP channel has not
> identified that the person on the receiving end has hung up.
>
> Is there a way of fixing this ?
On Thu, 8 Jul 2010, Danny Nicholas w
Thanks Zeeshan.that server is located at the headquaters and phones are
at different locations, even with default rfc2833 mode, other party IVR
prompts was not able to detect the tones, also 'Info' works good but not
with internal options like voicemail, etc. And inband is not being used as
we
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
Lyndon-Smith
Sent: Thursday, July 08, 2010 4:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Not detecting hang
We have had 20 calls over the last month where the SIP channel has not
identified that the person on the receiving end has hung up.
Is there a way of fixing this ?
TIA
Julian
--
_
-- Bandwidth and Colocation Provided by http:/
>From what you explained, it seems obvious that there exists some non-SIP
device somewhere in your communication path, and since voice is picked up as
DTMF, some device is also set to listen for inband DTMF.
What is the origination source of incoming calls to your system?
Zeeshan A Zakaria
--
ww
Sandesh,
Review the bug logs in particular ID: 0017571..there's a recent patch
that may apply to your issue.
http://issues.asterisk.org
Thanks,
Al
On Thu, Jul 8, 2010 at 4:18 PM, das sandesh wrote:
> Hi,
>
> We have few systems with asterisk 1.4.22.1 and we use sip trunking
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
--
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip trunking for them
not PRI's, one of our system is giving a problem of dtmf (rfc2833), like
when we dial the number that have IVR and enter the extension or access
code, it some time takes it and some times does'nt recognize the digits
d
On Thu, Jul 08, 2010 at 07:11:47PM +0200, Olivier wrote:
> 2010/7/8 Tzafrir Cohen
>
> > On Thu, Jul 08, 2010 at 05:48:53PM +0200, Olivier wrote:
> > > # lsmod | grep hfc
> > > hfc4s8s_l1 11716 0
> > > hisax 397700 1 hfc4s8s_l1
> >
> > I suspect you need to blacklist
VoIPInnovations from what I understand is pretty good, haven't dealt much
with them though. Worth a call and an interop.
--Matt Desbiens
//EOF
On Thu, Jul 8, 2010 at 3:33 PM, Adam Moffett wrote:
> I'm in the Northeast US and looking for any recommendations on Level3
> resellers. I don't do eno
I'm in the Northeast US and looking for any recommendations on Level3
resellers. I don't do enough volume to go to Level3 directly.
If there's anybody you'd definitely avoid I'd love to hear about that too.
Thanks,
Adam
--
_
Manmohan wrote:
> I was looking for audio conferencing solution where i got Web-meetme.
> I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working
> fine. I tried using Meetme even meetme app is working perfectly fine.
> I installed Webmeetme 4.0 and integrated with my asterisk. When
Please respond after the text which you are quoting and trim footers.
On Thursday 08 July 2010 13:23:18 Eric Hiller wrote:
> Tilghman Lesher wrote:
> > On Tuesday 06 July 2010 22:51:03 Eric Hiller wrote:
> > > Tilghman Lesher wrote:
> > > > On Monday 05 July 2010 19:17:00 Eric Hiller wrote:
> > >
Yes, I understand this part, but then what is the purpose of pollmailboxes if
not to trigger mwi?
> From: tles...@digium.com
> To: asterisk-users@lists.digium.com
> Date: Wed, 7 Jul 2010 01:23:34 -0500
> Subject: Re: [asterisk-users] Externnotify on pollmailboxes=yes
>
> On Tuesday 06 July 2010
Yes i agree; ok here the output of verbosity at level 3:
-- Executing [00212664800...@pstn2:1] GotoIf("SIP/100-081e3648",
"0?internal:external") in new stack
-- Goto (pstn2,00212664800450,2)
-- Executing [00212664800...@pstn2:2] Dial("SIP/100-081e3648",
"SIP/lo...@pstn2/011212664800450
2010/7/8 Tzafrir Cohen
> On Thu, Jul 08, 2010 at 05:48:53PM +0200, Olivier wrote:
> > Hi,
> >
> > I'm using Asterisk 1.6.1.18 with latest Dahdi Linux and Tools (rev 8854)
> and
> > libpri 1.4.10.2.
> >
> > During installation, I'm facing this :
> > # dahdi_genconf -v system
> > Default parameters
Marta Silva wrote:
> Hi All,
> I am trying to costumise the out number (not just the CallerID), for
> faxes sent from a specific sip client a fax machine connected to a
This was recently discussed on the iaxmodem mailing list. You can read
up on it here:
http://sourceforge.net/mailarchive/for
Hello guys,
I have this problem when a call is received in my PBX:
(Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) -->
(Internal Phone)
Reception works fine, but when conversation finishes and the agent at
internal phone hangs up, the call at caller's side is still alive for
m
On Thu, Jul 08, 2010 at 05:48:53PM +0200, Olivier wrote:
> Hi,
>
> I'm using Asterisk 1.6.1.18 with latest Dahdi Linux and Tools (rev 8854) and
> libpri 1.4.10.2.
>
> During installation, I'm facing this :
> # dahdi_genconf -v system
> Default parameters from /etc/dahdi/genconf_parameters
> Empty
On 7/8/10 5:07 AM, "Paul Belanger" wrote:
> On Wed, Jul 7, 2010 at 2:46 PM, Mike Ely wrote:
>> Maybe I missed something here? SIP users configured within Asterisk can
>> dial out just fine through the trunk. It's just when I try to use AMI that
>> it fails.
>>
> The far end is rejecting your
On Thu, Jul 8, 2010 at 8:30 AM, Jared Terrell wrote:
> # Span 1
> span=1,1,0,esf,b8zs
> bchan=1-23
> dchan=24
> echocanceller=mg2,1-23
> # Span 2
> span=2,2,0,esf,b8zs
> bchan=25-47
> dchan=48
> echocanceller=mg2,25-47
> # Span 3
> span=3,3,0,esf,b8zs
> bchan=49-71
> dchan=72
> echocanceller=mg2,4
I am trying to connect asterisk to EON millenium with SIP.
I am getting a 405 message back from EON
Looks like asterisk sends a register command , EON says OK
then asterisk sends sip options and EON says 405.
Is there a way not to send sip options?
THanks,
Jerry
--
__
Hi All,
I am trying to costumise the out number (not just the CallerID), for faxes
sent from a specific sip client a fax machine connected to a
Grandstream GXW-4004 V1.3B (I believe it uses the default number for my
PRI).
Can it be done?
Thank you very much in advance.
Regards,
- Marta
--
_
Hello,
using asterisk 1.4.30. No patch or anything else.
I just do the following in dialplan :
exten => 20,n,SIPAddHeader(Remote-Party-ID: "Testing"
)
When my Cisco calls my Grandstream, the name "Testing" appears on the
screen of my Cisco.
When my Grandstream calls my Cisco, only "20" app
Hi,
I'm using Asterisk 1.6.1.18 with latest Dahdi Linux and Tools (rev 8854) and
libpri 1.4.10.2.
During installation, I'm facing this :
# dahdi_genconf -v system
Default parameters from /etc/dahdi/genconf_parameters
Empty configuration -- no spans
Of course, I've got span descriptions in /etc/d
# Span 1
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
echocanceller=mg2,1-23
# Span 2
span=2,2,0,esf,b8zs
bchan=25-47
dchan=48
echocanceller=mg2,25-47
# Span 3
span=3,3,0,esf,b8zs
bchan=49-71
dchan=72
echocanceller=mg2,49-71
# Span 4
span=4,4,0,esf,b8zs
bchan=73-95
dchan=96
echocanceller=mg2,73-95
# Glo
On 07/08/2010 10:19 AM, Zeeshan Zakaria wrote:
> That's why I specifically mentioned Cat5 networks, because giga bit networks
> which use four pairs are called Cat6 networks.
>
> This is true that Cat5 networks are also used with gigabit hardware, but
> technically it is wrong. Cat6 hardware uses d
That's why I specifically mentioned Cat5 networks, because giga bit networks
which use four pairs are called Cat6 networks.
This is true that Cat5 networks are also used with gigabit hardware, but
technically it is wrong. Cat6 hardware uses different frequencies over
copper than Cat5, and mixing a
Zeeshan Zakaria writes:
> making use of the fact that both Cat5 networks and BRI ports
> don't use all the 8 pins, so why not use extra wires in the cable for
> something useful instead of wasting them.
For Ethernet, this is only true for 10Mbps and 100Mbps. Gigabit and up
uses all four pairs.
Hi,
I'm working on Asterisk and would like to use only Asterisk SIP signalling
for my Voip application.
I have written my own channel driver and want to integrate my own RTP with
Asterisk.
SIP signalling is working fine. But i could not find API's to get RTP Port
and IP address to start
witho
Hi all,
i do have the following setup
ISDN BRI Line -> openVOX Card/Asterisk 1.6.2.6/libpri 1.4.11.2 -> Dialplan
Dial DAHDI -> ISDN PBX -> ISDN Equipment
The user on the ISDN Equipment das enable call forwarding - Teilrufumleitung
/ Call deflection - so that call will get forwarded by the telco
Hello list.
I've been researching if there is a way of putting the recordings of
Mixmonitor in database (PostgreSQL or MySQL) in an automated way. I've read
that the native has voicemail in Asterisk via ODBC. And for the MixMonitor
has some way?
Someone on the list have it implemented?
Thanks,
R
On Thu, Jul 8, 2010 at 2:51 AM, Manmohan Singh Jandu
wrote:
> crashes giving segmentation fault.
>
Read doc/backtrace.txt on how to capture and generate a backtrace.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.
On Thu, Jul 8, 2010 at 6:29 AM, Adil Zaaraoui wrote:
> But it does not work.
> Any suggestion
>
Without posting a debug log it makes it hard to troubleshoot.
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
On Wed, Jul 7, 2010 at 2:46 PM, Mike Ely wrote:
> Maybe I missed something here? SIP users configured within Asterisk can
> dial out just fine through the trunk. It's just when I try to use AMI that
> it fails.
>
The far end is rejecting your call; SIP/2.0 401 Unauthorized.
If you can dialout w
Dear All,
I have "get full variable" AGI call to get the ANSWEREDTIME channel
variable. I have originated the call to one extension, once answered I have
called DeadAGI to control the call.
I have problem that after hangup the call AGI "GET FULL VARIABLE" returns
-1 for ANSWEREDTIME channel va
Hello,
Thank you for your reply.
Ok it works; now i have an other issue:
My asterisk(A) communicate with an other asterisk(B) via SIP protocole, my user
(U) communicate with asterisk(A) via IAX2 protocole; i tried as you said but
unfortunatly it does not work;
here is my configuration :
sip.
Hello list,
asterisk 1.4.30
2 situations in which call-limit should work, but it does not :
[Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The
device state of this queue member, test12, is still 'Not in Use' when it
probably should not be! Please check UPGRADE.txt for correct
On Thu, Jul 8, 2010 at 12:21 PM, Manmohan Singh Jandu
wrote:
> Hello Team,
>
> I was looking for audio conferencing solution where i got Web-meetme.
> I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working
> fine. I tried using Meetme even meetme app is working perfectly fine.
> I
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