On Tue, Feb 20, 2007 at 11:53:50PM -0600, Eric ManxPower Wieling wrote:
Tzafrir Cohen wrote:
obviously, the makefile used an incorrect kernel source tree to build
your systems. The package kernel-devel provides a partial kernel source
tree which is good enough for building modules (or at
Hi all!
I have Asterisk 1.2.14 (bristuff0.3.0preR8x) installed and i have 2 sip
accounts (A and B) registered at a sip-provider
I want my leds (functionkey 1) on the snom 190 to be lighted when a call
comes in on account A and my
functionkey 2 on account B.
Is there a way to do this with
Hi,
(Apologies for readers of Bristuff mailing as I already posted this message
to the list)
My setup is:
SIP hardphone - LAN --- Asterisk server
-ISDN
Asterisk server is :
Gentoo enabled with 1.0.8 bristuffed Asterisk
equipped with Junghanns Quad BRI with 2 BRI
My service provider only supports g729 and I tried what you have mentioned
here but same thing is happening here. Is there any why that I can see which
codec my service provider is pushing when I'm receiving call on my asterisk
server. When call comes comes to my server and then I type show g729
Thanks yusuf,
Any other experience on this subject? Anyone knows if Asterisk 1.4
already implement Radius authentication properly? Has anyone ever
patched Asterisk with the patch from the Digium Issue Tracker available
in the URL: http://bugs.digium.com/view.php?id=5424 and got well
Hi
I am writing a IVR app using phpagi and are coming up against a problem when
trying to detect DTMF. If I use the get_data function I dont seem to be able to
reliably detect 16 digits. If I try 10 digits then its fine but anything above
that seems to have a problem.
Any ideas anyone?
hello all,
i have a set up of 2 contexts with ivr features
and it works fine with voicemail also using callback=somecontext i can
callback
persons on that context
but problem is if i included third context i can only callback any one
context users
not all users
how can i solve this issue !
Dear Phil,
Thank you for your reply.
I have changed by extensions.conf as below.
And I also put my debug information for your reference.
It is a strange behavior. I got exited non-zero in it when I use ZAP channel.
If I use my SIP trunking gateway(outside), I got the return value is zero.
Hi All.
This is on Asterisk 1.2.13
I place a call between 2 SIP phones (with canreinvite=yes, qualify=yes).
I reset the phones (so they don't have time to say BYE).
Asterisk seems to think that the call is still ongoing. This persists
until I do a 'restart now'.
asterisk1*CLI show channels
Bump. Nothing heard.
On 2/19/07, Matt [EMAIL PROTECTED] wrote:
Greetings folks,
I'm currently dealing with a company to let me set Caller-ID-Name on
outbound calls. So far pretty happy with their services. The basic service
works like this:
* CLEC sets Point Code to point to this company
*
Is there any way, how to detect, what party starts touch monitor
recording? is some variable set?
I would like to deliver recorded file after call hangup to that user
using some shell script.
PJ
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In article [EMAIL PROTECTED],
Boris Bakchiev [EMAIL PROTECTED] wrote:
Hi,
Has anyone noticed degraded voice quality with HPEC?
I have a client running TE4XX card who configured HPEC for couple of
channels with echocancel=1024.
Whenever HPEC is used you get a background static in voice.
Hi Tony,
Its a dual core system and combined CPU usage was 2%.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: Thursday, 22 February 2007 12:07 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: The High
On Wed, Feb 21, 2007 at 01:06:36PM +, Tony Mountifield wrote:
Try using a utility like top to see what the CPU loading is with HPEC.
top itself can give you a pretty god idea, though it may be hard to
separate between zaptel itself and hpec.
oprofile can be handy. You'll probably need to
hi all, I'm trying to set up some iax2 trunks in Realtime architecture
with the same backend.
All work better (make call, receive etc etc) but when I do iax2 show
peers some asterisk don't show anything and other show the iax2 peers
but with status unknow.
Name/UsernameHost
Matt,
A Letter of Agency is almost always signed by the end subscriber and given
to the ILEC/CLEC. Its purpose is to allow someone other than the subscriber
(e.g. an Enhanced Service Provider or consultant) to make changes to, or
get information about, the customer's account (e.g. your account
Throughout the time I've been using * I've always made tests by calling
out on my SIP provider and calling my fixed line, it's often the only
way of getting an intelligent conversation :).
Since I've been trying trunk I find calls are being put on hold, I even
get music on hold on the calling
I am a direct subscriber to the CLEC. The DIDs have a SPID that belongs to
the CLEC, however the CLEC has given us full control of the numbers, and as
far as they are concerned, they are our numbers.
However, the CNAM company wants the CLEC to sign the LOA, instead of us.
On 2/21/07, Trevor
Ok,
I understand about the the wrong link, and I agree with that.
There are no kernel-devel-xen. I also tried with yum install
kernel-devel-xen and there was no match for that. But I found
That the right one is kernel-xen-devel, which I already finished installing.
I find out that there are no
Just out of interest: From former posts I understood that there is a
CALLERID service in US (for an extra fee, I assume) that gives both
number _and_ name of the caller...? I am aware of the fact that e.g.
EuroISDN lines can transmit alphanumeric callerid (and in fact I already
use that on an
On Wed, Feb 21, 2007 at 08:54:33AM -0500, Carlos Alperin wrote:
Ok,
I understand about the the wrong link, and I agree with that.
There are no kernel-devel-xen. I also tried with yum install
kernel-devel-xen and there was no match for that. But I found
That the right one is
Well,
I tried with the xen-devel and same result.
I tried changing the symlink to linux-2.6 -
/lib/modules/2.6.19-1.2911.fc6/build but that is incomplete since build
directory doesn't exists.
Also, I tried with no symlink and on each case:
[EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel
I removed Asterisk and reinstalled it from scratch. It seems to be working
now as module show like cdr now reports many more lines and now mentions
MySQL.
The database is the same as I didn't remove that, just the various files.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Benjamin Jacob wrote:
rfc2833 is the prefered way, as inband will work perfectly only with the
ulaw codec.
Out of curiosity, there is any 'document' about how RFC2833 would be
better or worse than SIP INFO ?
Pierre Marceau wrote:
Okay, in the SPA-941 admin I changed:
;DTMF Tx
agent monitoring screen?
curious,
which app are you using for that?
--
Chris Earle
Julian Lyndon-Smith [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Kyle Sexton wrote:
Started playing with 1.4 and I'm curious what uses people have come up
with for the Jabber integration? So
Anybody seen this behavior?
To determine if it's my config or a bug, could I trouble someone running
Asterisk 1.4.0 to set call-limit=5 on a moderately busy SIP interface as
a test? After a few hours a 'sip show inuse' should indicate the
interface is on calls that it isn't. The incorrect
Hi,
can someone give me a link to a howto about that?
I want to use jabbin with asterisk but dont find how to register jabbin
client in asterisk so it can make calls.
Thanks
Rodrigo
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Hello, list,
I am using Asterisk with an OpenLine4 card. It worked well with Asterisk
1.0. Then I upgraded the system, Asterisk 1.4 had some problem to
compile chan_vpb, but I managed to compile it manually. Still Asterisk
does not work because it refused to load chan_vpb module. I had to
Did you solved this Problem?
I have the same problem, and i can't solve it, did you know anything
about?
Thanks
Nico
On Thu, 14 Sep 2006, Kai Militzer wrote:
Hello everyone,
since some weeks I experience strange problems on my gateways to the
PSTN. The gateways use chan_ss7 and SIP. My
I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make
install and I don't see any errors. This is out of my modprobe.conf:
install tor2 /sbin/modprobe --ignore-install tor2 /sbin/ztcfg
install torisa /sbin/modprobe --ignore-install torisa /sbin/ztcfg
install wcusb
All calls through the system are being dropped when they are bridged
(Asterisk, Linux, pure VoIP system). The calling party here's half of
the word 'hello' for instance and the call is dropped.
I've noticed that hangup() was being called for some time now when the
call was bridged, but the
On Wed, Feb 21, 2007 at 09:52:32AM -0600, Mike Hammett wrote:
I go to my Zaptel 1.4.0 folder and run ./configure; make linux26; make
install and I don't see any errors. This is out of my modprobe.conf:
[ snip ]
However:
[EMAIL PROTECTED] zaptel-1.4.0]# modprobe zaptel
FATAL:
As a starting point for Linux installs I would recommend Ubuntu Linux.
Easy to setup, you don't need a Linux Administer degree to get started.
I stopped using Fedora after the 4th hard disk failure for no reason on
EXT 3.
PS
I too am an older developer. Let me know if you need
Hi,
thank You,
when i run zttool i get
Alarms Span â
â UNCONFIGUREDDigium Wildcard TE110P T1/E1 Card 0 â â
â UNCONFIGUREDZTDUMMY/1 1 â
Hello every body,
I ve installed Trixbox 1.2 on DELL OPTIPLEX GX 240, i upgreded it to
the latest version. I've 2 cards installed in the same pci channel via
a bridge or plug i don't know exactly what's his name( but a card (1
pci) that gives 2 pci channels)
the first card is TDM400P: it's ok
the
What asterisk version?
Bryan M. Johns
Partner
Shelton Johns Technology Group
Office: 678.248.2637
Direct: 678.229.1809
http://www.sheltonjohns.com
**Sent from my mobile phone**
-Original Message-
From: Jason Wolfe [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: 2/21/2007
The Asterisk and Zaptel development team has released version 1.2.14 of
Zaptel.
This release contains only minor changes, the most important of which
relates to single-port module support on Digium's TDM800P analog
interface card (previously these modules were not properly recognized by
the
Maybe Queue doesn't consider a SIP account that returns BUSY as in
use. That would be the only case where I could see needing call-limit.
James Fromm wrote:
We do the same thing only we use ringinuse=no and autopause=yes for the
queue. With autopause, if the agent is busy their interface in
On Wed, 2007-02-21 at 16:14 +, younss azzayani wrote:
Hi,
thank You,
when i run zttool i get
Alarms Span â
â UNCONFIGUREDDigium Wildcard TE110P T1/E1 Card 0 â â
â
Hi, there. I'm in the process of deploying one at a customer site
so I have a bit of experience with them. Set up of the unit is trivial...
You create a text file for the config and then use the provided uploader to
send the config to the unit. Because it *is* TDM, we went with a
Hi all,
Is there a way to keep track in Asterisk of which phones are online in
realtime using some MySQL DB table for exemple, much like sip show
peers does in the CLI?
Regards,
Ricardo.
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Hi,
I have similar symptoms (usually one-way audio like you, but sometimes
echoed, distorded, or low volume sound), in a simpler configuration,
using just SIP with a few phones and a TDM400 card with two FXOs:
Asterisk -- PSTN
I have kernel 2.6.18-XEN and using Asterisk 1.4
François.
Hi:
Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox server
isn't seeing the mainboard's APIC.
-Stephen-
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
1.2.1
Jason Wolfe, CTO
Click For A Call, Inc.
[EMAIL PROTECTED]
1-800-218-4951
o (770) 287-0273
c (770) 561-6956
This e-mail transmission may contain information that is proprietary,
privileged and/or confidential and is intended exclusively for the person(s) to
whom it is addressed. Any use,
Ricardo Carvalho wrote:
Is there a way to keep track in Asterisk of which phones are online in
realtime using some MySQL DB table for exemple, much like sip show
peers does in the CLI?
If you are using real realtime with rtupdate=yes in sip.conf
Asterisk stores the current time + sip
While trying to compile zaptel 1.2.8 on a FC5 I get the following error:
/lib/modules/2.6.19-1.2288.fc5smp/build
make -C /lib/modules/2.6.19-1.2288.fc5smp/build SUBDIRS=/usr/src/zaptel-1.2.8
modules
make[1]: Entering directory `/usr/src/kernels/2.6.19-1.2288.fc5-smp-i686'
CC [M]
On Wed, Feb 21, 2007 at 08:44:37PM +0200, Dovid B wrote:
While trying to compile zaptel 1.2.8 on a FC5 I get the following error:
/lib/modules/2.6.19-1.2288.fc5smp/build
make -C /lib/modules/2.6.19-1.2288.fc5smp/build SUBDIRS=/usr/src/zaptel-1.2.8
modules
make[1]: Entering directory
On Tue, Feb 20, 2007 at 12:08:15PM -0700, Natambu Obleton wrote:
Why not make it like DNS and have each provider have their lookups
deligated to a local server and then each ISP will run a caching
server that will use a serial number system to get updates.. just like
DNS.
I know there are
Hi guys,
I have a customer with asterisk registering 100 lines from my Voip Provider.
In some times during a day we receive this messages:
[Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to
schedule in the past?!?!
[Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime:
BT == Brad Templeton [EMAIL PROTECTED] writes:
BT Hey, we could even build a system where DNS can be used to take
BT any phone number and look up data about it, not just a name, but
BT even a URI to redirect calls to for it, a source of presence info
BT and more.
BT What a great idea!
I may be
What Asterisk command I can use to send a SIP INFO command? Thanks for
pointers.
Yuan Liu
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I have a sip.conf with stanzas for sip phones that have
'context=sip-incoming for some Grandstream phones and another stanza for
a Sipura SPA3000 with context=pstn-incoming.
Reviewing the code today, I was dismayed to see that all my outgoing
extens were mixed into those two. I have been
Hello list,
we are pleased ro announce that we have released a newer version of
QueueMetrics (1.3.3) that is able to monitor multiple Asterisk servers at
once, thus making it possible to monitor call centers running on clusters
or on high-availability configurations. See
On 2/21/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Hi:
Does Trixbox support
www.trixbox.org
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LA == Larry Alkoff [EMAIL PROTECTED] writes:
LA I have a sip.conf with stanzas for sip phones that have
LA 'context=sip-incoming for some Grandstream phones and another
LA stanza for a Sipura SPA3000 with context=pstn-incoming.
LA Reviewing the code today, I was dismayed to see that all my
LA
Lacy Moore - Aspendora wrote:
On 2/21/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Hi:
Does Trixbox support
www.trixbox.org
Thanks -- I know where the website is :P
Where did you think I got it?
-Stephen-
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I'm working on calls coming in to an asterisk box as H.323, and going out as
SIP to a remote device (a VoiceMaster). The remote device is refusing the
calls with SIP error 406 (Not Acceptable).
I have attached the SIP debug output below. It looks like codecs overlaps -
can anyone see why the
On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on FC5 x86_64
very good, but since FC keeps updating, I tried to follow newer kernel
versions.
If you want to save these hassles, why not use the packages bits that
are
Benny Amorsen wrote:
LA == Larry Alkoff [EMAIL PROTECTED] writes:
LA I have a sip.conf with stanzas for sip phones that have
LA 'context=sip-incoming for some Grandstream phones and another
LA stanza for a Sipura SPA3000 with context=pstn-incoming.
LA Reviewing the code today, I was dismayed
A client of mine has a Snom 320. Usually when he comes in each morning, it
is asking him for a password. A power cycle brings it back to normal
operation. How do I troubleshoot this further?
--Mike
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Mike,
A few things you can try, default administrator password should
by default. Maybe it just needs that entered.
Otherwise, if the phone is being used with Asterisk, there was a bug
on an issue like this which may have since been resolved, but non-the-
less is documented here:
Actually, 'context=' in sip.conf is the first place Asterisk looks for
when a number is dialled from the phone. It then uses 'includes' to
check for other options.
Usually, people use 'incoming' for their external lines, and something
else for the sip phones. I have used 'sip_phones' before.
Put your phones in the context=toll-access in sip.conf or zapata.conf
Put the phone lines in context=incoming in sip.conf or zapata.conf
extensions.conf:
[extensions]
exten = 667,1,Dial(SIP/whatever)
...
more exten lines to dial your phones here
[incoming] ; this is where calls from untrusted
You could also use Set(CALLERID(name)=1234*${CALLERID(name)})
and then on the other astereisk server use app_cut to reformat CID and
extract the Var.
On 2/20/07, Eric Bishop [EMAIL PROTECTED] wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except
Stephen Bosch wrote on Wednesday, February 21, 2007 12:26 PM:
Hi:
Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox
server isn't seeing the mainboard's APIC.
TB is really CentOS 4.4, which is really RHEL 4.4.
Now all you have to do is find out if RHEL supports it. :)
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 21 Feb 2007, at 23:06, Axel Thimm wrote:
On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on
FC5 x86_64
very good, but since FC keeps updating, I tried to follow
Hello Eric.
I don't fully understand your example.
I _think_ you have in extensions.conf:
[incoming]
include = extensions
[extensions]
exten = 667
more exten here
[toll-trunks]
exten = 91NXXNXX
more exten here
[toll-access]
include = extensions
include = toll-trunks
My understanding of
Larry Alkoff wrote:
Hello Eric.
I don't fully understand your example.
I _think_ you have in extensions.conf:
[incoming]
include = extensions
[extensions]
exten = 667
more exten here
[toll-trunks]
exten = 91NXXNXX
more exten here
[toll-access]
include = extensions
include = toll-trunks
this is my zaptel.conf::
[EMAIL PROTECTED] ~]# cat /etc/zaptel.conf
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module loading order
# Span 1: WCT1/0 Digium
genzaptel is _not_ your friend when setting up E1.
PaulH
On Thu, 2007-02-22 at 00:46 +, younss azzayani wrote:
this is my zaptel.conf::
[EMAIL PROTECTED] ~]# cat /etc/zaptel.conf
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This
Hi All,
Just wanted to share a story:
We turned up a new customer yesterday evening, typical situation, Cisco 2600
Router with T1 PRI card pointed to the customer's analog PBX with 2 data
T1's linked back to our network. The router PRI was configured as a gateway
on our CCM 4, like we've
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Hello Eric.
I don't fully understand your example.
I _think_ you have in extensions.conf:
[incoming]
include = extensions
[extensions]
exten = 667
more exten here
[toll-trunks]
exten = 91NXXNXX
more exten here
[toll-access]
include =
Larry Alkoff wrote:
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Hello Eric.
I don't fully understand your example.
I _think_ you have in extensions.conf:
[incoming]
include = extensions
[extensions]
exten = 667
more exten here
[toll-trunks]
exten = 91NXXNXX
more exten here
Axel,
Thanks for your advice, but as I tried to found the real problem overpass
the search is just like close my eyes.
I'm trying to learn in order to not repeat same mistake twice. I don't know
how the rpm's are build, and I don't think that
You can apply on every kind of variation you can
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Hello Eric.
I don't fully understand your example.
I _think_ you have in extensions.conf:
[incoming]
include = extensions
[extensions]
exten = 667
more exten here
[toll-trunks]
exten =
Larry Alkoff wrote:
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Hello Eric.
I don't fully understand your example.
I _think_ you have in extensions.conf:
[incoming]
include = extensions
[extensions]
exten = 667
more exten here
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 21, 2007 9:15 PM
Subject: Re: [asterisk-users] Problem Installing Zaptel
On Wed, Feb 21, 2007 at 08:44:37PM +0200, Dovid B wrote:
While trying to compile zaptel
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Eric ManxPower Wieling wrote:
Larry Alkoff wrote:
Hello Eric.
I don't fully understand your example.
I _think_ you have in extensions.conf:
[incoming]
include = extensions
[extensions]
Glad to hear you had a workaround.
I would suggest re-queing your TAC case, perhaps you got a outsourced or less
experienced engineer at Cisco. Their support has varied depending on which
city/group you get. Some have more experience then others.
While your 2600 from 2001 timeframe it should
Could anybody give me an authoritative answer on whether Asterisk can
support T.38 pass-through when the clients are behind NAT? We have
Asterisk servicing clients behind NAT (with nat=route, canreinvite=no)
and would love to get T.38 going but have had no luck so far. The
following case:
I have one Asterisk box registering to another via SIP and on the registar
console I keep getting:
-- Got SIP response 603 Declined (no dialog) back from xxx.xxx.xxx.xx
Anyone know how to turn off this feature?
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[EMAIL PROTECTED] wrote:
Stephen Bosch wrote on Wednesday, February 21, 2007 12:26 PM:
Hi:
Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox
server isn't seeing the mainboard's APIC.
TB is really CentOS 4.4, which is really RHEL 4.4.
Now all you have to do is find out
Ray wrote:
Could anybody give me an authoritative answer on whether
Asterisk can support T.38 pass-through when the clients
are behind NAT? We have Asterisk servicing clients behind
NAT (with nat=route, canreinvite=no) and would love to get
T.38 going but have had no luck so far. The
Hi Richard,
there was a thread regarding this a while ago on the dev list which resulted
in a patch being made to allow variable passing via IAX2 channels. See
http://bugs.digium.com/view.php?id=7619 for the patch which I think is in
SVN or anyhow, is not in 1.2
I have recently backported
On Thu, Feb 22, 2007 at 11:58:06AM +1100, Paul Hales wrote:
genzaptel is _not_ your friend when setting up E1.
/usr/local/sbin/genzaptelconf that comes with trixbox: no. It is very
old copy of genzaptelconf.
try xpp/utils/genzaptelconf for something that has supported E1 for
quite a while
On Thu, Feb 22, 2007 at 12:52:22AM +0100, Jens Vagelpohl wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 21 Feb 2007, at 23:06, Axel Thimm wrote:
On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on
Dear Phil,
The extension 'h' was a great idea although I still got the error
exited non-zero.
Thank you for your help.
Best regards,
Charles
2007/2/21, Phil Reynolds [EMAIL PROTECTED]:
Quoting Charles Wang [EMAIL PROTECTED]:
Dear Phil,
Thank you for your reply.
I have changed by
On Wed, Feb 21, 2007 at 08:35:12PM -0500, Carlos Alperin wrote:
Axel,
Thanks for your advice, but as I tried to found the real problem overpass
the search is just like close my eyes.
I'm trying to learn in order to not repeat same mistake twice. I don't know
how the rpm's are build, and I
LA == Larry Alkoff [EMAIL PROTECTED] writes:
LA If it's not a security issue I might as well have all phones with
LA context=default in sip.conf even though voip-info specifically
LA warns against that. Wonder why?
Random SIP calls from the internet could end up in context default, if
that is
MH == Mike Hammett [EMAIL PROTECTED] writes:
MH A client of mine has a Snom 320. Usually when he comes in each
MH morning, it is asking him for a password. A power cycle brings it
MH back to normal operation. How do I troubleshoot this further?
It isn't necessary to power cycle, it's enough to
On Thu, Feb 22, 2007 at 12:52:22AM +0100, Jens Vagelpohl wrote:
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On 21 Feb 2007, at 23:06, Axel Thimm wrote:
On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
I tried to test Asterisk 1.4 on FC6 x86_64. I have it working on
BTW Carlos, all your posts are with Importance: High in this thread.
On Wed, Feb 21, 2007 at 08:35:12PM -0500, Carlos Alperin wrote:
Axel,
Thanks for your advice, but as I tried to found the real problem overpass
the search is just like close my eyes.
Not really what I was suggesting, see
**I have one Asterisk box registering to another via SIP and on the registar
**console I keep getting:
**
**-- Got SIP response 603 Declined (no dialog) back from xxx.xxx.xxx.xx
**
**Anyone know how to turn off this feature?
Look at:
On Thu, Feb 22, 2007 at 07:47:18AM +0100, Axel Thimm wrote:
On Thu, Feb 22, 2007 at 12:52:22AM +0100, Jens Vagelpohl wrote:
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On 21 Feb 2007, at 23:06, Axel Thimm wrote:
On Tue, Feb 20, 2007 at 06:01:35AM -0500, Carlos Alperin wrote:
Surely there must be a simpler way than patching the Asterisk code? After
all this is Asterisk-to-Asterisk registration not some third party
softswitch idiosyncrasy. Would setting up fake voicemail boxes help?
On 2/22/07, Davy Chan [EMAIL PROTECTED] wrote:
**I have one Asterisk box
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