This sounds like an asterisk @ home issue. Not an asterisk issue.
Asterisk at home uses a GUI that limits what asterisk can do, look at the
config files it creates in (/etc/asterisk) and voip-info.org
Michael
On Mon, 21 Mar 2005, David Shaw wrote:
Hello, I tried to add an outside number (my
You don't get it
We have to work harder to outshine Skype . : )
Its not like ford and GM. They do too very different things...we don't
have to outshine them we already do.
On Sat, 26 Mar 2005, Stephen wrote:
Hi All,
Thanks for all the comments and opinions.
I think in terms of features and
Please use the Asterisk at home forms at
http://sourceforge.net/forum/?group_id=123387
For you asterisk at home help.
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Please seek help from the amp people as this list is for asterisk.
On Wed, 30 Mar 2005, hank smith wrote:
hello I have asterisk 1.0 running on fedora core3 and amp version 1.06 I think
is the version its the version down below the current release, I have fwd
working threw iax on outbound calls
On Sat, 2 Apr 2005, ruben cuevas rumin wrote:
Hi all,
I'm a Telecomunication Engeenering student. I have to develop a VoIP
apliccation using SIP protocol. I have to develop the SIP Server, and
the SIP clients.
I think I can use Asterisk for this issue. I have installed it and I
have run it, but I
We would like to know if there is a way to broadcast (in realtime) a
conferance. We hold large phone conferances
and would like to know if we could have some of our users listen over a
streaming services. Formats we have looked at include: Shoutcast,Real
Networks,QuickTime, and dare I say
company
Mike
-Original Message-
From: nathan [mailto:[EMAIL PROTECTED]
Sent: 28 May 2003 11:18
To: asterisk-users
Subject: [Asterisk-Users] VOIP phone suppliers in the UK?
Hi All,
Can anyone recommend a supplier and/or a particular a model of
SIP voip phones in the UK? I don't
.
Their incessant complaint (the police, that is) is that
it's too expensive to roll out much beyond a trial system.
Driving the hardware cost down would just leave them
their revenue costs to whinge about.
Mike Pellatt
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'
and nothing in the logs.
Can someone help me with this? Thanks,
Mike
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not getting either a ring or a no route to
destination error. It's as if Asterisk is trying to reach the phone for
the full 15 seconds, and only then giving up.
My tests are done with a Polycom 650 phone, if that matters (I doubt it
does). I've seem the same behavior on Polycom 501 and 320.
Mike
?
Is this any other obvious option that escapes me?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith
Sent: Wednesday, August 01, 2007 14:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem
available
again after the configured number of milliseconds? Or will it be considered
unreachable until the next register attempt by the device?
Regards,
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Cennami
Sent: Wednesday, August 01, 2007 17:56
have
PoE enabled.
From the switch to our test phone, we have a typical blue RJ-45 cable, going
into the special PoE-RJ45 cable Polycom provides with the 501. And then
that cable into the phone.
What the heck could be wrong in such a simple setup?
Mike
(hint_reg=${EXTEN}-reg}
exten = _XXX,hint,SIP/${hint_reg}
exten = _XXX,SIP/${EXTEN}-reg}
Or, even easier (if it can even be done) is write a function:
exten = _XXX,hint,SIP/ReturnCorrectRegistration()
What's the best way to approach my problem?
Mike
be done in the same Asterisk
priority. See my previous email for background (Buddy watch and the hint
priority - brain teaser).
Any help is extremely appreciated.
Mike
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(SIP/${A}) ; I need to know ${A} first, but I can't
know before this line is called (it's very DB driven).
What can I do? Am I dead in the water here?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, August 08, 2007
)})) , what I don't know is
how to actually write the function with a return value (and Googling this
doesn't get me any relevant result, apparently).
I'd be most thankful for some link to a page that shows how to write such a
function in Asterisk.
Mike
-Original Message-
From: [EMAIL PROTECTED
)})
In the hope of getting to see Noop(Hello World) in my CLI, I get the
following Asterisk error:
Aug 8 13:40:48 ERROR[5771]: pbx.c:1402 ast_func_read: Function AGI not
registered
AGI certainly seems registered as it worked in the first case. Again,
something obvious I missed?
Thank you,
Mike
then actually integrating that code in
larger project...unfortunately.
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
FitzGibbon
Sent: Wednesday, August 08, 2007 14:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
?
Mike
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to firmware
2.x and get whatever benefits you can get from that.
Regards,
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen
Sent: Thursday, August 09, 2007 10:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
:11187 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from xx.xxx.xx.xx, but there is
no hint for that extension
Wellthere is! Is there any way I can do this?
Mike
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I feared so, but I have already started working on this. Thanks for the
confirmation.
Too bad, the rest of my design was relatively elegant (IMO) and easily to
modify.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent
subscribecontext (one word) is another attribute of a peer (sip.conf). I am
using it as part of a MYSQL table that holds all my sip registrations, and
that works fine. I did have to add the column, since it wasn't part of the
table construct that can be found on the wiki.
Mike
Possibly NAT related issues. Try to add the line qualify=yes to your SIP
peer/friend/user.
I just discovered that, wonderful little gizmo.
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Lengua
Sent: Thursday, August 09, 2007 16:13
To: asterisk-users
In the interest of making things cleaner, I'd like to know if I can just
reload one single conf file. Let's say I have two files, extensions.conf
which includes small_file.conf.
I only want small_file.conf reloaded, not the main file. Is this at all
possible?
Mike
The thing is that I make them automagically reload from outside Asterisk (by
calling asterisk -rx extensions reload)
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: Friday, August 10, 2007 10:32
To: Asterisk Users Mailing List
that mattered as opposed to the whole thing. For
all I know, this could be triggered while I am coding some new thing and
could screw up my dialplan.
But I guess I won't be doing this.
Regards,
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken Williams
Sent
this? Because the %*$%/$ hint fonctionnality can't
accommodate variables fetched from a DB like the rest of my dialplan.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: Friday, August 10, 2007 12:11
To: Asterisk Users Mailing List - Non
that they can see status by looking at the line icon,
this will only confuse them).
Second question, can you set up the phone so that this status, which is
shown in the line icons, is also shown in the contact directory?
Regards,
Mike
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of very
obvious typos/spelling mistakes.
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen
Sent: Friday, August 10, 2007 10:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom question - removing
debug and doesn't work.
using RFC2833 (AVT) and application/dtmf-relay does the same as above.
Mike
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is
not that I want.
How can I make sure that only the external leg is counted?
Mike
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I am having a similar issue with 4.0.0. Mine is that it doesn't get any
DHCP address (gets stuck waiting for an address).
I fixed it by going back one to the previous bootrom version, worked like a
charm.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
I use a 650, so YMMV, but it's working with mine.
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al lists
Sent: Wednesday, September 26, 2007 01:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes
hey Folks,
Just curious if anyone has suggestions on how one can get a near
FREE(I hope) DID number.
I am experimenting with asterisk, for home use.
thanks,
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disable
colorization from the command line, and I did try using nocolor=no in the
config files. No luck.
Regards,
Mike
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?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mik Cheez
Sent: Wednesday, April 09, 2008 19:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Jumped from 1.2.7 to 1.4.19,
missing CLI colors
Hi,
I have a big issue during transfers (using Polycom phones, but I don't think
that's relevent) with Asterisk 1.14.19. Basically, the value contained in
${CDR(accountcode)} dissapears.
Here is the relevant code snippet:
--
exten =
Thanks, that`s what I ended up doing. Still, it doesn't seem to be WAD,
since the CDR(accountcode) is correct and suddently dissapears.
Is this a bug (I was looking through the bug system and couldnt match this
with a bug, but then again I am not a developer) or is it really WAD?
Mike
Hi,
I`m trying to use Chanspy for a customer that wants to listen to his
employees so he can train them better (or so he claims). In any case, it
looks simple but there is something I`m not doing right.
When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234)
When I use, on another
Hi,
(me again, my upgrade to 1.4 is more painful then I imagined it would be).
I just noticed that the command show hints shows all hints correctly, but
none of them ever are InUse (even if I use a line and dial out) like I used
to on 1.2.
Can`t find a bug in the bug tracking system, is
. And the good thing is you often do not have to do anything but set
the upload bandwidth (yes there is an automatic mode, but it's not that
great).
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Graves
Sent: Thursday, April 17, 2008 10
My own Chanspy(g(GROUPNAME)) works 2 times out of three (roughly). The
other time, it crashes Asterisk. Using 1.4.19 too.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Rawlings
Sent: Thursday, April 17, 2008 14:10
To: Asterisk Users
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Michelson
Sent: Thursday, April 17, 2008 17:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Chanspy on Asterisk 1.4.19
Mike wrote:
My own
when the phone is ringing/on a call. Asterisk doesn't support all those
fancy status that you can select from the phone.
Mike
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Anthony,
What bug report ID# would that be? Not being a dev I find it hard to know
which of the 4 chanspy bug I need a patch for, since none of them seem to
refer to a 1.4.19 bug.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anthony
config looks good.
Regards,
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Lee, John (Sydney)
Sent: Friday, April 18, 2008 08:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Newbie Polycom
(accountcode)})
And the CLI shows :
-- Executing [EMAIL PROTECTED]:2] Set(SIP/00041234432-1-b7d4b908,
GROUP=internal-5149070849) in new stack
Which seems right. But it never shows up in the CLI when checking for group
channels.
Any clue? Did something major change between 1.2 and 1.4?
Mike
Wow that was easy. Thanks! My wrong syntax must have worked in 1.2 by pure
chance, or I must have erased the brackets by mistake while I was tweaking
the config files following the upgrade.
Thanks alot, you saved me alot of time and grief.
Regards,
Mike
_
From: [EMAIL PROTECTED
Hi,
I used to have ## configured in asterisk 1.2 for blindxfer. Now, when I
press ## I hear it on the other end instead of initiating a transfer. What
has change and how can I go back to the old behavior? I kept the same
feature.conf file with these lines:
[featuremap]
blindxfer = ##
If you want to avoid a mecanical lifter, the only option I know of is a
Jabra GN9350 with a Polycom EHS (electronic hookswitch) cable.
It came out only a month ago I believe. I use one, and wouldn't give it up
for the world.
Regards,
Mike
-Original Message-
From: [EMAIL PROTECTED
Hi,
What is the syntax to set more than one variable in the SIP.conf file for a
particular sip peer? (using the setvar line)
Regards,
Mick
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Hi,
I am having trouble with Polycom forwards and Asterisk. Basically, I have
no clue on how to force callerid or even custom variables (set using SetVar
in the sip.conf file) on the transfered call.
For example, I set a variable called var_a to foo. When the call comes
in, the variable
using realtime SIP entries) and not in the diaplan per say.
My setvar column is this:
internal_callerid=blabla 123;did=551234
I tried adding underscores before did (as in: internal_callerid=blabla
123;__did=551234) but that didn't help.
Mike
Did you try _var_a? Iirc you need to prepend it with an underscore to
make the variable persistent.
Forget my previous email, it didn't quite work that simply but I tweaked my
dialplan and you had the right solution.
Thank you,
Mike
Subject: Re: [asterisk-users] Trouble with Polycom phones
Yes, I was using a name instead of an IP address. And if memory
servesI *think* it is using TCPprefered...but I could be wrong.
Kevin
Mike wrote:
I have been running into a few issues with Asterisk/polycom and I am
running
Hi,
I have what I think is a relatively advanced question. Any help is
appreciated, even if it's not a complete answer.
I am using Asterisk in mostly realtime fashion, specifically SIP
registrations are in a MySQL table. This works fine (mostly). I also set a
few variables in the setvar
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk : using setvar with IP Realtime and
variable inheritance
Mike wrote:
If I hardcode this value in my dialplan using two underscores before it
(i.e
Setvar(__did=551234) ) this works. But I can't
Hi,
I`m occassionally seeing CDR(accountcode)'s value empty at a place in my
diaplan where it was filled with some value a few lines before, with nothing
else having changed it.
It`s giving me headaches (as I rely on it for MySQL queries). Anything I
can do?
Mick
at 2:21 PM, Mike [EMAIL PROTECTED] wrote:
Hi,
I`m occassionally seeing CDR(accountcode)'s value empty at a place in my
diaplan where it was filled with some value a few lines before, with
nothing
else having changed it.
It`s giving me headaches (as I rely on it for MySQL
See, to get back to your answer, this is what I`m not understanding:
Again, this works fine. The problem is when I forward my calls to
another
outside line (using Polyocm phones), and need to know the ${did} value
at
that point. It's empty.
Right, so the call path is:
Provider --
I get that a lot since moving to 1.4.21 (from 1.4.18 or something).
[Jun 17 09:19:54] WARNING[22053]: res_config_mysql.c:360 update_mysql: MySQL
RealTime: Failed to query database. Check debug for more info.
Question 1: what debug file should I be looking at?
Mick
Just an addition: that happens big time when I do a sip reload from the
CLI
I know this should help me already, but it doesn`tÂ…
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, June 17, 2008 09:23
To: 'Asterisk Users Mailing List - Non-Commercial
Hi,
I am in a weird situation where a variable seemed ignored, but not always.
That variable is __TRANSFER_CONTEXT.
Basically, I have a phone registered with asterisk. It's context is
internal. Outgoing calls go through that context (all good).
When I get an incoming call which I
the
call initiated using the Dial g option is hung up ?
Regards,
Mike
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asterisk-users
Doug,
Thanks for the quick answer. How does that help me though, since this is a
per channel variable and not a global variable?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, October 07, 2008 16:54
To: Asterisk Users
. I've played around with the kewlstart and
loop-start setting but without knowing what the line is going to do,
it's difficult to know how to configure Asterisk.
Does anyone have any experience of Telewest?
Thanks,
Mike.
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On Fri, Oct 10, 2008 at 08:10:39AM +1930, Luis Morales wrote:
Mike,
Can you tell us :
- asterisk version
- zaptel version
When you call over this line, when you hangup did you hear an busy
tone ? or any class tone ? To do this test connect your lines to
analog phone and make a call
On Sun, Oct 12, 2008 at 11:18:40AM +0100, Gordon Henderson wrote:
On Thu, 9 Oct 2008, Mike wrote:
I'm guessing this lamp is on an ordinary analogue phone you have?
Yeah, this is a bog standard 9 quid analogue phone.
OK. A bit convoluted this as I'm not local to the PBX, but an IAX trunk
in advance,
Mike
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What country are you in? This is a truly global marketplace and mailing
list. We have people from the UK, Ireland, Oztrailia, New Zealand,
Bolivia, Russia, China, India, Argentina, etc. All over the world, really.
Saying what country you need the DID/DDI in will narrow it down somewhat.
I am
can do to make my other phones ring?
Thanks,
Mike.
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(new in 2.x)
Let me know if that worked.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones
Sent: Monday, April 16, 2007 09:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 501 issue
for
that matter) into a WiFi phone?
Mike
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, where do I find an adapter for NA power that turns into 2V 5A
DC current?
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Friday, April 27, 2007 13:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best
Hi,
I've been asked for a headset recommandation for Polycom SoundPoint IP
phones. Since I believe they use a pretty standard headset jack (correct me
if I am wrong) it's really a general recommandation on headset.
Regards,
Mike
in
the same SIP entry?
Mike
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Yeah ok. That doesn't help.
What I mean is I want a call to go out on ProviderA, UNLESS it's down and
then go to ProviderB.
I want it to ring 30 seconds and then Hangup if nobody has answers.
I DON'T want to dial both, only one or the other.
Mike
-Original Message-
From: [EMAIL
of
programming with every line numbered like BASIC
Can you easily mix and match AEL and standard Asterisk (i.e. my old code
with new code I would put in?)
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall
Sent: Friday, May 11, 2007 22:22
Hi,
I just got a Polycom 330 and, of course, I don't have the firmware and
sip.cfg files to provision it. Where can I get those?
Mike
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the number
doesn't increased as planned, it stays at 1 channel used.
Regards,
Mike
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) twice.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, February 27, 2007 10:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Do I understand GROUPs correctly?
Mike wrote:
Hi
into a Set(group) again to increment channel before dialing a
cell phone using a dial(cellphone#) cmd.
If that doesn't work, how do I accomplish the same kind of thing elegantly?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent
Thank you, that is exactly what I needed.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp
Sent: Tuesday, February 27, 2007 11:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Do I understand
Jason,
If you do test if JR's tip works, please share your finding with us. I am
interested in this as well.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Thursday, March 01, 2007 21:11
To: asterisk-users@lists.digium.com
polycom phones that I wanted to reboot)?
Thanks,
Mike
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Thanks Dave, good info!
And thanks to those who confirmed I needed to write a script because there
were no built in functions, I appreciate that info too.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton
Sent: Monday, March 12, 2007
values must be identical. Is there any way to
change that?
Mike
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!
Is this true? And if so, what happens when the Phone doesn't connect
directly to the switch? (let`s say there is wiring in the wall that goes to
a patch panel, for example. Do I need to change all the wiring in the
office?)
Mike
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Thanks for all the replies, this definitely helps me!
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Wednesday, March 28, 2007 12:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PoE
a message waiting?
Regards,
Mike
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I`ve been noticing alot of those messages in the CLI lately:
Apr 5 11:18:02 NOTICE[25593]: chan_sip.c:6444 check_auth: stale nonce
received from 'sip:[EMAIL PROTECTED]
I haven't changed my configuration in ages. What could be the cause of this
suddent appearance?
Mike
Thanks David and Chris, appreciate the response
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton
Sent: Thursday, April 05, 2007 11:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom
useful.
Also, I read that the phone offers TLS security. What does that mean? I
understand Asterisk does not, but is this something that could be possible
with futur asterisk developement?
Mike
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Ah, thanks. I didn't realize this.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Nickerson
Sent: Monday, April 09, 2007 16:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom 330/320
in a scenario where you reuse traditional phones
to connect to SIP servers, but can they accomodate my scenario? And if so,
what line of ATA should I be looking at?
Mike
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asterisk-users mailing
Probably, if I only needed one FXO. What is the customer has 4 channels
(PSTN lines)? Don't I need 4 FXO?
And, about the Sipura, it looks like it would do what I want, but it only
has one FXO, limiting it's usefulness.
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
Thanks Alex,
That was my original thought, to just buy a TDM400 from Digium and put in as
many FXO as I wanted, but I liked having the ease of just buying something
off the shelf, even if it meant paying a little more.
But it looks like I won't have much of a choice.
Mike
-Original Message
for them to react. I've
downgraded to 1.6.7, and the problem dissapeared.
I can't imagine I'm the only one having that issue, and that issue was also
present in 2.0.1 for me.
Did anybody else have this problem? What did you do to fix it? Am I stuck
with 1.6.7 forever?
Mike
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