Hi,
Using MeetMee(1234,M) on asterisk 1.6.2.9 (debian/testing) we have
terrible sound: the MOH is unrecognizable and speakers can't be
understood; it sounds ghostly. However the prompts (your are the only
one in this conference, etc.) sound fine.
Our server has a Digium T410P card with two E1
On Tue, Feb 08, 2011 at 10:59:47AM -0600, Danny Nicholas wrote:
Any idea?
I use mpg123 to play my MOH so I can control the volume (my users complain
that standard MOH is a bit loud).
Forgot to add that our MOH sounds fine when listened to (on the same
extension as MeetMe) with
On Tue, Feb 08, 2011 at 11:09:19AM -0600, Warren Selby wrote:
On Tue, Feb 8, 2011 at 11:04 AM, Louis-David Mitterrand
vindex+lists-asterisk-us...@apartia.org wrote:
Forgot to add that our MOH sounds fine when listened to (on the same
extension as MeetMe) with MusicOnHold(default). So it's
Hi,
Using asterisk 1.6.2.0 I have a queue definition with strategy=linear.
How do I jump to the next dialplan item after having tried
(unsuccessfully) all queue members?
If I use Queue(test,n) then only the first member is contacted. And if I
omit the n option then all members are retried
Hi,
What is the best channel driver to use asterisk 1.4.x with a 4BRI isdn
card from Beronet or Junghanns (same hardware, different pcid)?
Are these cards now supported by plain (non-patched) dahdi/zaptel
modules?
Thanks,
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Hi,
After upgrading a debian/lenny server to 1.4.26 I get this error:
== Manager 'munin' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'munin' logged on from 127.0.0.1
[Jul 26 17:45:12] ERROR[12354]: utils.c:966
On Fri, Jul 24, 2009 at 11:14:47AM +0200, Philipp Kempgen wrote:
Louis-David Mitterrand schrieb:
On Fri, Jul 24, 2009 at 10:37:38AM +0200, Michiel van Baak wrote:
On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3
Hi,
I'd like to disable MWI on certain lines of my IP650 Polycom phone. So I
removed the mailbox= parameter from that line's peer section in
sip.conf. Yet the envelope still appears in front of that line and the
phone MWI keeps blinking.
Where should I look to completely disable MWI on a certain
Hi,
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten = 2131/,n,Playback(no_unknow_callerid_here)
exten = 2131/,n,Hangup
And now, after upgrading to 1.6.1.x it matches every callerid.
Did something change?
Thanks,
On Fri, Jul 24, 2009 at 10:37:38AM +0200, Michiel van Baak wrote:
On 10:17, Fri 24 Jul 09, Louis-David Mitterrand wrote:
This used to work fine in 1.4:
exten = 2131/,1,NoOp(reject3: ${CALLERID(num)})
exten = 2131/,n,Playback(no_unknow_callerid_here)
exten = 2131/,n
Hi,
Is there a way on Polycom phones to show an agent whether he is logged
in or not?
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On Wed, Jun 17, 2009 at 01:08:33PM -0500, Danny Nicholas wrote:
If this is a recorded sound, you might want to truncate it with lame or
audacity. It is quite common in my shop as we record using the phones.
Thanks for this suggestion.
The problem was indeed a silence at the beginning of my
Hi,
I have a 2/3 second gap between the end of a welcome message played with
Playback and the start of the Queue music. Here is the dialplan:
exten = ${EXTEN},1,NoOp($EXTEN)
exten = ${EXTEN},n,SIPAddHeader(Alert-Info: Ring_CCC)
exten =
Hi,
After upgrading to 1.6.x and hdvoice (g722) polycome phones I am
wondering how to optimize asterisk sounds and music on hold to take
advantage of that codec. I often listen to a special music extension on
my headset:
/usr/bin/wget -q -O - http://music.example.com | /usr/bin/madplay -Q -z -o
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
Thanks,
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On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote:
Louis-David Mitterrand schrieb:
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
Digium's BRI cards are also based on Cologne Chip - thus you could try
Digiums BRI
On Tue, Jun 09, 2009 at 04:04:29PM +0300, Tzafrir Cohen wrote:
On Tue, Jun 09, 2009 at 02:45:26PM +0200, Klaus Darilion wrote:
Louis-David Mitterrand schrieb:
Hi,
Can I use a simple HFC PCI (Cologne chip) card with asterisk 1.6.x ?
What drivers are available?
mISDN
Hi,
Is anyone here using OrderlyStats with asterisk in a call center
setting? If so what what is your experience with it? Is that software
really free for asterisk users?
Or is there a better option out there?
Thanks,
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On Mon, May 04, 2009 at 10:04:53PM +1000, Rob Hillis wrote:
Louis-David Mitterrand wrote:
Hi,
Is anyone here using OrderlyStats with asterisk in a call center
setting? If so what what is your experience with it? Is that software
really free for asterisk users?
Or is there a better
On Tue, Feb 10, 2009 at 01:56:16PM -0800, Mik Cheez wrote:
I use them both; my legacy dialplan is all .conf and new stuff is .ael.
I find AEL to be the better option when jumping around, but that's
just my opinion.
But isn't AEL just converted into .conf language anyway? Or has this
Hi,
How hard is it to integrate asterisk with Microsoft CRM?
Thanks for any suggestions, pointers, etc.
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On Wed, Jan 21, 2009 at 09:02:51AM -0200, David fire wrote:
how hard is to integrate whit a virus?
sorry
ok i read MS CRM but... did you tried VTiger? www.vtiger.com the next
release (5.1) will be integrated whit asterisk not only click to dial and
popups on incoming calls a queue monitor
On Wed, Jan 21, 2009 at 12:58:51PM -, Andrew Thomas wrote:
Try http://forums.vtiger.com/viewtopic.php?t=14314
Thanks, this is a really interesting link.
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On Wed, Jan 21, 2009 at 09:00:41AM -0500, Jon Weisman wrote:
ok what about people that have no choice but to use MS CRM?
That's also my concern, as MS CRM is my customer's choice, not ours, and
I may or may not succeed in steering them toward an open-source solution
such as vTiger. They already
Hi,
I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but
then my TE410P alarms stay RED and no zap channels can be created, even
if they are correctly listed by zap show channels. I tried adding
dahdichanname = no to asterisk.conf's [options] to no effect.
Going back to
On Tue, Nov 11, 2008 at 09:49:14AM +0100, Louis-David Mitterrand wrote:
Hi,
I tried upgrading from debian's 1.4.21.2 package to vanilla 1.4.22 but
then my TE410P alarms stay RED and no zap channels can be created, even
if they are correctly listed by zap show channels. I tried adding
On Thu, Nov 06, 2008 at 11:46:48AM +0100, Louis-David Mitterrand wrote:
Hi,
After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we
experience crashes at random intervals with:
[Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame
read(0, unfinished
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process:
midget packet received (1 of 4
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
Your monitoring app is not sending valid IAX2 packets to the
server. If
it was sending a true IAX2 POKE, it would be a valid packet and
wouldn't
generate this warning.
Could asterisk at least _not_ report this harmless,
On Sat, Nov 08, 2008 at 02:33:18PM +1100, Rob Hillis wrote:
Tzafrir Cohen wrote:
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
I'd take this warning seriously. It means that your monitoring app isn't
monitoring what you think it is.
I always want to know when I get
Hi,
After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we
experience crashes at random intervals with:
[Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame
read(0, unfinished ...
+++ killed by SIGSEGV (core dumped) +++
Process 15755 detached
On a second
Hi,
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process:
midget packet received (1 of 4 min)
This is triggered by the monitoring app sending a POKE to the iax port.
The warning appears
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process:
midget packet received (1 of 4
On Wed, Oct 15, 2008 at 11:30:49AM -0500, Tilghman Lesher wrote:
On Wednesday 15 October 2008 10:26:50 Louis-David Mitterrand wrote:
For some calls (usally telemarketers) entering through a BRI zap channel
I somtimes notice the callerid on my polycom 601 phone and the CDR's
'src' field
Hi,
Using asterisk 1.4.21.2.
For some calls (usally telemarketers) entering through a BRI zap channel
I somtimes notice the callerid on my polycom 601 phone and the CDR's
'src' field don't match. They are even totally different. And the
displayed callerid is nowhere to be seen in the CDR record.
Hello and sorry for the OT,
Is it possible for a wireless headset of which the base is connected to
a Polycom IP601 to remotely answer a call? In the same way as a
bluetooth headset.
thanks,
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Hi,
Returning to my office I find two missed calls (from autodialers) that
my IP601 displays as originating from 011. However the CDR
database recorded the call this way:
calldate: 2008-04-04 14:18:16+02
clid: 0172752780
src:
On Thu, Feb 28, 2008 at 11:10:37AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
zenon:~# module-assistant -t build zaptel
make[3]: Entering directory `/usr/src/linux-2.6.24.3'
scripts/Makefile.build:46: *** CFLAGS was changed in
/usr/src/modules/zaptel
Hi,
I am trying to build zaptel 1.4.8 with kernel 2.6.24 on debian/sid:
zenon:~# module-assistant -t build zaptel
make[3]: Entering directory `/usr/src/linux-2.6.24.3'
scripts/Makefile.build:46: *** CFLAGS was changed in
/usr/src/modules/zaptel/Makefile. Fix it to use
On Wed, Oct 31, 2007 at 04:53:49PM +0400, Arun Kumar wrote:
try to reduce number of calls on trunk or create multiple trunks.
The flood happens when I have only one call on the trunk.
On 10/31/07, Louis-David Mitterrand [EMAIL PROTECTED]
wrote:
Hi,
Using 1.4.13 and trunking a single
Hi,
Using 1.4.13 and trunking a single iax channel to a similar box my
asterisk console is flooded with:
[Oct 31 10:49:34] WARNING[5195] chan_iax2.c: Maximum trunk data space
exceeded to xx.xx.xx.xx:4569
Known issue?
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Hi,
Using 1.4.13 is it possible to ignore 302 redirects from sip devices
belonging to a queue?
For a queue that rings the whole office it doesn't seem very useful to
obey a redirect programmed on a phone.
It seems this was the default behaviour in 1.2.
Thanks,
On Wed, Oct 31, 2007 at 06:11:47PM +0100, Louis-David Mitterrand wrote:
Hi,
Using 1.4.13 is it possible to ignore 302 redirects from sip devices
belonging to a queue?
For a queue that rings the whole office it doesn't seem very useful to
obey a redirect programmed on a phone
Hello,
I'm trying to upgrade a Thomson ST2030 phone froms its default 1.42
firmware to the latest version (1.56) through tftp.
The phone loads the .inf file, then the correct firmware file (as stated
in the ST2030S.inf), then it reboots and loops doing these same things
again and again. The
Hi,
a block of my extensions.conf no longer works after upgrading from
1.2.17 to 1.4.4. I have:
[macro-dialout]
exten = s,1,Gosub(s-${ARG1},1)
exten = s,n,Congestion
;; default
exten = _s-!,1,Gosub(s-NET,1)
When calling that macro whith no argument
On Wed, Apr 18, 2007 at 01:04:31PM +0200, Tim Koehler wrote:
Hi,
are you using PoE or power supplies?
As power supllies usually are not grounded it could be that it's comming
from the power source.
We are using PoE
You could try using a grounded PoE switch or probably a power backup to
On Mon, Apr 16, 2007 at 12:25:55PM +0200, Bas van der Veen wrote:
Hello,
Did you find anything while testing the LAN? Also, can you confirm that
switching the switch, cabling, etc. did NOT solve the problem?
It did not.
We finally changed the server itself and reinstalled from a
Hello,
We are at wit's end on this. One (and only one) of our five asterisk
installation is giving us real headaches. Buzzing and/or choppy sound
interfere with conversations. I recorded some conversations with
monitor() and no problem whatsoever appear in the recording, while the
local user
On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote:
Yes, I recently saw this with a 501, in my case the network drop was
the problem. If you have a good tester then run it on the connection.
I had another drop near by and just swicthed to it.
Was that phone using POE ?
Hello,
After upgrading my kernel to mISDN-1.1.1 while keeping asterisk-1.2.16 I
no longer match any extension. Apparently the dad is empty. However I
can see the number just before it (146472130):
P[ 4] I IND :SETUP oad:!?145201798p
¡146472130 dad:
¡146472130 pid:2
Hi,
At one location we have a user whose Polycom IP430 suffers from erratic
reboots. We swapped his phone for a brand new one, but the problem
remains.
Has anyone seen that?
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On Wed, Mar 21, 2007 at 07:07:00AM -0400, joe a. wrote:
Did you swap the power module as well? If POE, did you swap the
patch cord?
If the power module plugs into a power strip did you change that? or
at least the position in the strip?
Thanks for the tought, but the IP430 has no
On Wed, Mar 21, 2007 at 04:07:34AM -0700, Henry Cobb wrote:
On 3/21/07, Louis-David Mitterrand
[EMAIL PROTECTED] wrote:
Hi,
At one location we have a user whose Polycom IP430 suffers from erratic
reboots. We swapped his phone for a brand new one, but the problem
remains.
Has anyone seen
On Wed, Mar 21, 2007 at 05:53:27AM -0500, Bruce Reeves wrote:
Yes, I recently saw this with a 501, in my case the network drop was
the problem. If you have a good tester then run it on the connection.
I had another drop near by and just swicthed to it.
What kind of test tool would you suggest?
Hello,
I'm using the classic [stdexten-macro] in extensions.conf whereby a call
is picked up by voicemail after a certain ringing time.
When programming a SIP phone to redirect calls (SIP 302 redirect) to
another extension I'd like to avoid that voicemail pickup so that the
call goes into the
On Tue, Mar 06, 2007 at 07:18:08AM -0600, Eric ManxPower Wieling wrote:
How can I detect that a call has been redirected and should no longer be
intercepted by vm?
That should happen by default. The call should get sent to the new
place and it should act like the call was directly
On Fri, Jan 12, 2007 at 02:33:54PM -0500, Doug Crompton wrote:
I am using spa3000 hardware - 2.0.1(5673) firmware - 3.1.3(GWa)
I have used newer firmwares but find that 3.1.3 had less echo problems.
Thanks again Doug for that detailed explanation.
As for the DTMF playback level and DTMF
Hello,
Before throwing in the towel with my Sipura 3000 has anyone had much
success with that adapter connected to a door phone?
In our setup a doorphone is connected to the SPA's fxs port. When a
visitor rings, asterisk calls a group of Polycoms and the person who
answers has to enter *1 to
On Fri, Jan 12, 2007 at 10:58:16AM -0500, Doug Crompton wrote:
The spa3000 does not play well with Asterisk with dtmf rfc2833 signaling.
Set BOTH the sip.conf AND the spa3000 to inband for DTMF. That would be
the line1 tab on spa3000. This applies to the fxo (pstn) also if you are
using it for
Hello,
When a user forwards his SIP phone to another extension (say an absent
boss to his secretary) I'd like the unanswsered forwarded call to end up
in the new destination's voicemail. With my current diaplan the call is
handled by the original recipient's voicemail:
On Tue, Dec 05, 2006 at 08:02:35AM -0600, Eric ManxPower Wieling wrote:
Louis-David Mitterrand wrote:
Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's
unreliable and perfectly good hosts will become UNREACHABLE for no
apparent reason, while SIP connections keep going
Hello,
We have several regional asterisk's connected to a central one making
the the PRI calls through a TE410P card.
When using SetCallingPres(prohibited) on a call at the regional level,
that setting it not forwarded to the central asterisk and the call is
made as if no callerid had been
On Thu, Nov 30, 2006 at 08:52:50AM -0600, DM wrote:
Setup:
Office A:
router: Linksys WRT54GS running SVEASOFT Alchemy-pre7a v3.37.6.8sv
Asterisk: v.1.2.4
static IP
Office B:
router: Linksys WRT54GL running Linksys firmware v4.30.2
Asterisk: v.1.2.7.1
dynamic IP (using dyndns name)
Hello,
I am trying to use chan_misdn on a junghanns QuadBRI card.
Using the latest install-misdn-mqueue from beronet, all installation
went well apparently. However when I try to load the card it is not
recognized:
# modprobe hfcmulti type=0x04 protocol=0x12,0x12,0x2,0x2
On Wed, Nov 29, 2006 at 11:45:50AM +0100, Louis-David Mitterrand wrote:
Hello,
I am trying to use chan_misdn on a junghanns QuadBRI card.
Using the latest install-misdn-mqueue from beronet, all installation
went well apparently. However when I try to load the card it is not
recognized
Hello,
With the following setup:
- asterisk 1.2.13,
- zaptel 1.2.10
- bristuff 0.3.0-PRE-1v
- quadbri card,
after a few hours of normal operation incoming calls suddenly fail to
enter with the following message:
received SETUP message for call that is not a new call
restarting asterisk
On Mon, Nov 27, 2006 at 09:44:08AM +0200, Kevin Boddy wrote:
I've got a few 8 port Junghanns BRI ISDN cards. Dialling in and out is
working fine but the Telco's busy or invalid number indications are not
being passed through to the user. I have priindication=passthrough in my
zapata.conf but
Hello,
Here is our setup:
asterisk-A --LAN-- nat-router --Internet-- asterisk-B
A and B have appropriate friend entries in their sip.conf with a
qualify=yes.
The router forwards anything on sip,iax and sip/rtp ports to A.
The problem: SIP/A remains UNREACHABLE for SIP/B, however A sees B. No
Hello,
Using 1.2.13 with bristuff:
exten = 8599,1,Answer()
exten = 8599,n,Wait(1)
exten = 8599,n,MusicOnHold(default)
Whan the call comes through a zap (telco) channel I can't hear the
music, but through a sip/iax channels I hear it.
Any idea why?
Thanks,
Hello,
here is our layout:
asterisk-A --- WAN --- asterisk-HQ --- WAN --- asterisk-B
calls are routed with SIP between asterisk's (found IAX to unreliable).
When asterisk-HQ attempts to native-bridge OR simply forward calls
between A and B no sound is sent.
If either leg (A - HQ or
On Mon, Nov 06, 2006 at 06:47:01PM -0600, Eric ManxPower Wieling wrote:
Louis-David Mitterrand wrote:
Hello,
Using a PRI (E1) with the euroisdn protocol, I don't seem to get any
specific message from the telco when attempting to dial a non-existent
number. Asterisk returns a busy
Hello,
On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an
unalocated number? I always get 3 (no route) which is less than helpful.
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Hello,
Using a PRI (E1) with the euroisdn protocol, I don't seem to get any
specific message from the telco when attempting to dial a non-existent
number. Asterisk returns a busy/congested code, but nothing indicating
the number's real status.
How do you guys manage that issue? Do you record
Hello,
Our Polycom's 601 can no longer register or communicate with the asterisk
server when using kernel 2.6.18.x. Cisco 79XX and other phones still
work though.
Downgrading back to latest 2.6.17.x solves the problem for Polycoms, but
I'd really like to understand what's going on there...
On Fri, Oct 27, 2006 at 12:15:15PM -0400, Doug Lytle wrote:
Louis-David Mitterrand wrote:
Hello,
Our Polycom's 601 can no longer register or communicate with the asterisk
server when using kernel 2.6.18.x. Cisco 79XX and other phones still
work though.
I'm running just 2.6.18 fine
On Fri, Oct 27, 2006 at 05:11:24PM +0200, Louis-David Mitterrand wrote:
Our Polycom's 601 can no longer register or communicate with the asterisk
server when using kernel 2.6.18.x. Cisco 79XX and other phones still
work though.
Downgrading back to latest 2.6.17.x solves the problem
Hello,
When I restart asterisk the cisco 7960/7940 phones (sip fw 7.5) fail to
re-register themselves with asterisk, even though I put
timer_register_expires: 60 in SIPDefault.cnf
Is there a way to have these phones register themselves every 60
seconds?
Alternatively, can asterisk be made
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote:
That's a bug with the 7.5 firmware. I would suggest upgrading to the
8.4 version, we've been running it for a few weeks in a test environment
and everyone's been pretty satisfied with the new firmware (read:
nobody's complained).
Hi Kape,
With latest asterisk 1.2.12.1, zaptel 1.2.9.1 and bristuff 0.3.1s after
a while calls become stuck: either the caller or callee can't hear the
other party, or heavy static is heard. An asterisk restart fixes it for
a short while only.
This doesn't happen with our older installs
Hello,
Since our telco messed with our PRI in some way, we get corrupt faxes
like these:
http://zenon.apartia.fr/stuff/corrupt_fax.pdf
We use the lastest asterisk with a TE410P and spandsp.
(for some strange reason, our neighbour company has a traditional pbx
fed by 7 BRI's and sees the same
Hello,
We have a TE410P connected to an EuroISDN E1 with these span
definitions:
span=1,1,0,ccs,hdb3
span=2,1,0,ccs,hdb3
span=3,1,0,ccs,hdb3
span=4,1,0,ccs,hdb3
Why should we add crc4 to these definitions? What does it do?
Thanks,
Hello,
Has anyone noticed that the Polycom IP430 has a low incoming/outgoing
sound level?
Is it a firmware issue or should I adjust my zap's tx/rxgain?
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Hello,
I am looking for the latest 1.6.7 Polycom firmware?
Is it available somewhere?
Thanks,
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On Tue, Aug 08, 2006 at 11:42:01AM -0500, Eric ManxPower Wieling wrote:
Louis-David Mitterrand wrote:
Hello,
I am looking for the latest 1.6.7 Polycom firmware?
Is it available somewhere?
What issues are you experiencing that 1.6.7 fixes?
Flaky buddy watch with 1.6.6
Hello,
Using 1.2.9.1 with bristuff and a QuadBRI card, phantom/zombie
channels accumulate throughout the day and end up blocking all incoming
calls.
It's the first time we have this problem and several similar
installations work fine.
We suspect bad cabling between the telco and the QuadBRI
Hello,
I've got a few Cisco phones to maintain and need access to firmware
files. Dealers here in .fr want unreasonable prices for a Smartnet
subscription.
Where can I get a better deal on the Net?
Thanks,
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Hello,
I have a TE410P connected to a telco on port1 and legacy Matra pbx on
port2.
When calling an extension managed by the legacy pbx through the telco (with a
normal pots phone), I get ringing. However when calling that same extension
through a SIP phone, no ringing is heard.
Here is the
On Thu, Jun 08, 2006 at 02:04:43PM -0500, Andres wrote:
We are currently considering the Linksys POE switch for a small
Asterisk office deployment. There will be no separate wiring closet
to put it in. Can anybody tell me if this switch has a loud fan?
Yes, this switch is loud. It only
On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote:
Well, these are encouraging words :)
You're basically telling me that I should tell my client to buy other
phones. I agree that you cannot compare these phones with Cisco or
Polycom. After all, like you said, what do you
On Wed, Jun 07, 2006 at 08:27:28AM -0400, Daniel Salama wrote:
While I would agree with you, the price difference between a GXP-2000
and a Polycom 430 or a Thomson ST-2030. These latter units are, at
least, twice as expensive as the GXP-2000.
BTW, I never heard of the Thomson ST-2030,
rue Lamartine
78000 Versailles
France,
fadge
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Louis-David
Mitterrand
Sent: 07 June 2006 13:36
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: GXP-2000 (steer clear)
On Wed, Jun
Hello,
At a client site yesterday I installed a dozen GrandStream GXP-2000's
with 1.1.0.13 firmware but I had to backtrack and reactivate the old PBX
and phones: network access for users windoze PC's through the phone's
switch port was unbearably slow, making it almost impossible to work.
Hello,
I have a Gigaset S44 connected to a quadBRI NT port. Receiving calls
works phone, however when dialing out from the phone the call is dropped
to the 's' extension, as if no extension had been dialed:
-- Accepting voice call from '492389990' to 's' on channel 0/2, span 4
On Mon, May 22, 2006 at 12:25:34PM +0200, picciuX wrote:
for provisioning files to be taken, you have to change the config_sn
parameter each time you modify the file, otherwise the phone assumes nothing
has changed.
Even after a factory reset of the phone? (ie: power-cycle with
speaker+mute
On Mon, May 22, 2006 at 10:11:30AM -0500, [EMAIL PROTECTED] wrote:
I'm going to try and lay out all the relevant information I have here
in this one post. I can provide more info if necessary.
ISSUE 1:
Office A routinely looses connection to Office B. When typing IAX2
Show Peers, it will
Hello,
After reading all the docs and going through the menus, I still can't
find the voicemail access button or menu sequence on the ST2030
(http://www.voip-info.org/wiki/view/Thomson+ST2030)
Also I can't get phone provisionning through tftp to work. Configuration
files are loaded but the
On Thu, May 04, 2006 at 12:51:52PM -0700, Tom Engleward wrote:
--- Vahan Yerkanian [EMAIL PROTECTED] wrote:
Andrew Kohlsmith wrote:
On Thursday 04 May 2006 11:31, Louis-David
Mitterrand wrote:
I've got this low-ping 100%-up dsl connection
between two asterisk
1.2.7.1 servers
I've got this low-ping 100%-up dsl connection between two asterisk
1.2.7.1 servers. And oftentimes one of them would declare its opposite
UNREACHABLE.
Why can this happen? The host stanzas in iax.conf have raw IP's, so no
DNS monkey business here.. An inquiring mind wants to know.
On Thu, May 04, 2006 at 10:31:17PM +0500, Vahan Yerkanian wrote:
Andrew Kohlsmith wrote:
On Thursday 04 May 2006 11:31, Louis-David Mitterrand wrote:
I've got this low-ping 100%-up dsl connection between two asterisk
1.2.7.1 servers. And oftentimes one of them would declare its opposite
Hello,
I have several asterisk 1.2.7.1 servers connected through iax2 and often
the local asterisk would no longer see the remote one, even thought the
link is high quality and the ping is perfect.
Is there some issues to take into account about IAX2 connections?
Is asterisk's DNS resolution
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