Hello,
I just received what seems to be a nice SIP-DECT gateway but can't
make it work with asterisk. The manual is very unclear (written in
chinese english) and the web configurator is ambiguous as well.
Has anyone succeeded in making one of these babies work with * ?
info:
Hello,
I am about to put an asterisk server between the telco E1 and our old
Matra PBX.
Should I use an ethernet cross cable? Something else?
Thanks,
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On Sat, Apr 22, 2006 at 08:09:13AM -0700, Paul Mahler wrote:
A T carrier cable is not the same as an ethernet cable. A T carrier
cable uses a real metal shielded RJ-45 and loosely twisted pair wire.
With most modern T carrier equipment, you can use a CAT-5 ethernet
cable instead of a
On Sat, Apr 22, 2006 at 11:59:21AM -0400, Alexander Lopez wrote:
Can't anyone stop self-promotion and tell the poor guy what he needs.
A T1/E1 X-over cable using an RJ-45 (8-cond.) is pinned out as follows:
1 - 4
2 - 5
3 - NU
4 - 1
5 - 2
6 - NU
7 - NU
8 - NU
NU = Not Used
I
On Mon, Apr 24, 2006 at 07:21:18AM +0800, Leo Ann Boon wrote:
Louis-David Mitterrand wrote:snip
Should I use a T1 cross cable to connect the telco's socket to the
TE410P card?
When I tried straight cat5 cables, both leds remained red at each end.
However this E1 socket works fine
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
On Wed, May 03, 2006 at 07:48:37AM -0500, Rich Adamson wrote:
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
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
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
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 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,
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
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.
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
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
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
Hi,
I looked at the docs and probably missed it: is there a way to set a
dialplan on the GXP-2000? (to avoid having to press Send)
Thanks,
--
Computers are useless. They can only give answers. - Pablo Picasso
___
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Hi,
I'm trying to get a SPA-3000 to work with a Siemens Gigaset 3010 DECT
(cordless) phone. I tried every localization scheme I could find on the
Net, including the settings recommended by the Voxilla wizard.
This Gigaset works fine and rings when plugged directly into the telco's
analog phone
Hi,
The gxp-2000's lack of a dialplan (or did I miss it?) led me to activate
its early dial option to avoid pressing Send after dialing. Thus the
dialplan is controlled by asterisk.
It creates an extension matching problem:
exten = _00[1-9].,1,Macro(dialcapi)
If I dial 0012 the extension is
On Fri, Nov 18, 2005 at 03:30:32PM +0100, Leif Neland wrote:
The gxp-2000's lack of a dialplan (or did I miss it?) led me to
activate its early dial option to avoid pressing Send after
dialing. Thus the dialplan is controlled by asterisk.
It creates an extension matching problem:
exten =
Hi,
I've recently reinstalled a Diva in my asterisk server (alongside a
QuadBRI :-) to test the nice features Armin has been adding in
chan_capi.
The capi.conf format has changed, so my question is how do I define a
deflect= statement for different incoming MSN's?
I've tried to define a section
On Fri, Sep 30, 2005 at 01:11:19PM +0200, Armin Schindler wrote:
Also, is there a way to detect that a SIP phone has an active forward
number and capi-deflect any incoming calls to that number?
If you can retrieve this information from extensions.conf, then you can use
my example above.
On Tue, Sep 07, 2004 at 02:24:06PM +0100, Nick Barnes wrote:
Hi all,
I have just received the following e-mail from an Asterisk user:
I just made a call via BT to a mobile. Then an incoming call came in and
Ann else answered it - it made my call go completely fuzzy and I could hear
what
On Wed, Aug 25, 2004 at 04:22:26PM -0700, [EMAIL PROTECTED] wrote:
I finally have my 7920 working though I'm seeing this bizarre
behavior. As soon as the 7920 boots and authenticates with the AP my 7960
release's its ip.
Hi,
I have exactly the same problem. Have you found a solution or
Hello,
I am trying to convert my hint priorities from the old style:
exten = 2130,hint,SIP/0146472130
to Asterisk Extension Language (AEL) style.
I haven't found anything in the docs, wiki or examples about it.
How should I do it?
--
Sigs have been known to cause cancer in California.
On Fri, Dec 02, 2005 at 12:05:08PM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
to Asterisk Extension Language (AEL) style.
I haven't found anything in the docs, wiki or examples about it.
I don't believe hints are supported in AEL at this time.
Thanks for the heads-up
Hello,
Using * 1.2.1 with chan_capi CVS on a Diva server I am mostly happy.
However when a phone redirects a call (user forward) and all ISDN
channels are busy, the call goes out through an IAX connection and it
takes a few seconds to get a ring state from the remote * server. This
makes the
Hello,
I just received a couple SX440isdn phones and was wondering if they can
be plugged into a Diva 4BRI port in NT mode and work with
asterisk+chan_capi?
I know they probably work fine with mutliHFC cards with either bristuff
of chan_misdn but since I have some spare Divas, I was curious
On Wed, Dec 28, 2005 at 04:02:00PM -0800, William Boehlke wrote:
The 830s are nice but limited because they do RAID on a card and have but
one suitable PCI slot. So you can have an interface card or RAID, but not
both.
Linux software raid is, in our experience, much better than any hardware
On Mon, Jan 02, 2006 at 11:25:02AM +0800, Craig Guy wrote:
Are you using raid for performance or redundancy? Software raid is
nice except when the drive that fails is the one with your boot
partition on it. I guess you could always tftp boot the kernel or
something.
On our raid1 machines
On Mon, Jan 02, 2006 at 01:01:44PM -0800, Mike Fedyk wrote:
Administrator TOOTAI wrote:
Craig Guy a écrit :
Are you using raid for performance or redundancy? Software raid is nice
except when the drive that fails is the one with your boot partition on it.
I
guess you could always
On Fri, Dec 30, 2005 at 10:23:26PM +0100, Armin Schindler wrote:
On Fri, 30 Dec 2005, Louis-David Mitterrand wrote:
Hello,
I just received a couple SX440isdn phones and was wondering if they can
be plugged into a Diva 4BRI port in NT mode and work with
asterisk+chan_capi?
Yes, I
On Mon, Jan 09, 2006 at 03:50:55PM +0100, Armin Schindler wrote:
On Mon, 9 Jan 2006, Louis-David Mitterrand wrote:
I am now using a cross cable and the green led lights up on the Diva
port when plugging the phone in.
When dialing from the phone I get no debug or trace at the asterisk
On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
[C:4] 22:0188:202 - D-X(003) 02 01 7F
[C:4] 22:0189:202 - D-X(003) 02 01 7F
[C:4] 22:0190:202 - D-X(003) 02 01 7F
[C:4] 22:0191:201 - MDL-ERROR(G)
[C:4
On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote:
On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
[C:4] 22:0188:202 - D-X(003) 02 01 7F
[C:4] 22:0189:202 - D-X(003) 02 01 7F
[C:4] 22:0190
On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote:
On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
[C:4] 22:0188:202 - D-X(003) 02 01 7F
[C:4] 22:0189:202 - D-X(003) 02 01 7F
[C:4] 22:0190
On Fri, Jan 13, 2006 at 02:03:20PM +0100, Armin Schindler wrote:
On Wed, 11 Jan 2006, Louis-David Mitterrand wrote:
On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote:
On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
On Tue, 10 Jan 2006, Louis-David
Hello,
Using asterisk-1.2.1 I am trying to convert my music-on-hold files from
.wav to alaw:
% sox moh.wav -r 8000 -c 1 moh.al resample -ql
The file sounds fine when listened with:
% sox mox.al -t ossdsp /dev/dsp
But when listened through asterisk with an alaw SIP phone the
On Tue, Jan 17, 2006 at 06:07:27PM +0100, Karsten Wemheuer wrote:
Did I forget something in my conversion command?
Are You using bristuff 0.3.0-PRE-1f?
Yes.
I've had the same issue. Dan Austin
wrote a notice in a mail on this list, which solved the problem.
Configure the following
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,
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 am looking for the latest 1.6.7 Polycom firmware?
Is it available somewhere?
Thanks,
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To UNSUBSCRIBE or update options visit:
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,
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,
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,
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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To
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 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,
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,
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
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)
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
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:
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
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
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
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,
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
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,
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
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,
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,
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
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