ne, I need it to all 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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n => 12345,hint,Dial(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
Koh
(ooga)})) , 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-
Fr
ious I missed?
Thank you,
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: Wednesday, August 08, 2007 12:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to write a function with a return value in
ing C code 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: [aste
or me? Is there some deeper meaning to this
notice I am getting?
Mike
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ld allow you to move 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-Commercia
]: chan_sip.c: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.
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
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 Use
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 Wil
? 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
I want removed (it's
good enough for me 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 show
recise than that and specified
"extension reload foo.conf"
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Friday, August 10, 2007 09:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-
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 a
t; there.
using RFC2833 (AVT) and application/hook-flash shows nothing on console
using sip debug and doesn't work.
using RFC2833 (AVT) and application/dtmf-relay does the same as above.
Mike
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ich 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 PROTECTE
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 c
"asterisk". Which is NOT what I
want.
Is there a standard way to say "hid my number"?
Mike
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oud be hidden unless it's
a call to 911 or something.
What we have is a SIP connection, not a PRI, is there anyway to do something
like that with SIP? Would that be provider-specific?
Mike
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t when the
call initiated using the Dial g option is hung up ?
Regards,
Mike
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
aste
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
e
clear down occured. 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.
_
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
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
I would be interested in knowing if this can be changed. It can`t have been
designed like this with no option to change it.
So I`m throwing this question back in the arena: Can you get the Polycom 501
to ring when a calls comes in and the user is already on a call?
Mike
-Original Message
Thanks, I know your right (I tried the second option). Problem is that the
phone doesn`t RING. The light flashes, the as far as an audio ring goes,
it`s completely silent.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Vincent
(C)
Sent
Thanks. That's an ok solution. I just thought I could make the Polycom
ring normally (or even better, with decreased volume) when a new call comes
in.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones
Sent: August 3, 2006 11:00
Hi,
I`m trying to record a conference, and I`ve been using .wav format to get
decent audio quality. The conference goes fine, but when I listen to the
recording after, I hear horrible echo (which I couldnt hear on the conf
call itself).
Whats causing this?
Mike
one phone.
Is this the
case?
Mike
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n I make it call with 102?
3) In call lists, my
numbers are listed as 555-555-. Yet my asterisk dial plan requires me
(by design) to press 9 first. How can I make the phone put the 9 by
itself?
Thank you for any
help you may give me,
Mike
___
-
I was expecting a more elegant answer to the "9 to dial out" problem with
the Polycom 501. Sure I can change my dialplan, but that means I have to
adapt my dialplan to the phone, while the opposite seems like the way to go.
Thanks for the answer,
Mike
-Original Message-
Fr
Title: RE: [asterisk-users] Polycom 501 config questions
Pretty much like Doug said: because people expect
it.
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k.
CreasySent: August 31, 2006 11:45 AMTo: Asterisk Users
Mailing List - Non-Commercial Discussion
m 501. Sure I can change my dialplan, but that means
I have toadapt my dialplan to the phone, while the opposite seems like the
way to go.Thanks for the answer,Mike-Original
Message-From: [EMAIL PROTECTED][
mailto:[EMAIL PROTECTED]] On Behalf Of Jerry
JonesSent: August 30, 20
.mike
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perfect !
thank you very much !
On Fri, 2006-09-01 at 09:12 +0200, Giorgio Incantalupo wrote:
> Hi Mike,
> yes we have one and it is working good.
>
>
> Giorgio Incantalupo
>
>
> mike wrote:
> > Dear list,
> > it is possible to have one quadbri (with o
Hi
all,
I just found out how
to set the column userfield, in the CDR DB to whatever I
desired. Can I add multiple custom columns to the DB and fill them from
the dialplan, or is it limited to one column?
I am using Asterisk
1.2.4 and MYSQL for the CDR DB.
Mike
ere is no real issues keep
me from doing so efficiently.
Regards,
Mike
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. What setting
could cause this? I have a fairly fancy dialplan, but I havent changed anything
else than the diaplan. All system-wide Asterisk settings are default as
far as I know.
Thanks,
Mike
___
--Bandwidth and Colocation provided by
o find out what I can from my limited RTP expertise. I appreciate
the response and the hints.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Mountifield
Sent: September 7, 2006 12:09 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk
uot;never"
BTW, Someone mentionned how Unlimitel (one of my VoIP
providers) has great support and I agree. I just know they can`t help me because
the issue is between my PBX and my SIP phone, not with them.
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Colli
)
Is there a
workaround? Or am I forced to use UserField, which is already used for
something else (and using the src field in CDR would really be ideal for
me)
Mike
___
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aste
nsion
567 as a valid try and let Asterisk handle the error ?(instead of the phone
trying to do what it think is best and handling the error on it's
own).
Is there an Asterisk
setting for that?
Failing that, is
there a Polycom setting to disable this "intelligent&q
hone tries the send something (I can see an icon moving on the phone)
but the phone stays quiet (no stuttering tone or whatever). It waits, I can
input more digits on the phone.
Let's just take 1) and 2). Why is Asterisk not going into the i extension
like it should?
Mike
-Original Mess
ould match a pattern
(_9X) with a few more digits and so waiting for those digits from the
user? How can I disable this or turn it off? The Polycom 501 "supports 484
responses", but how can I either:
1) Disable it in the phone
2) Disable it in Asterisk
Mike
Using INVITE req
ial
9-555-55- and then press "send".
Am I wrong? Cause did try the above example, and I got a 484 response
back...
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
"ManxPower" Wieling
Sent: September 8, 2006 5:49 PM
To: As
t a number, the phone company does
wait a few seconds for the last digit. But there is a timeout, and
eventually I get a fast busy. That`s what I want. And apparently, I can`t
get that.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
"Man
That`s the only relevant thing I haven`t yet found how to do.
Regards,
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton
Sent: September 8, 2006 3:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [as
Here it is:
When I dial 845, I get fast busy. When I dial 9-555-555-, it dials
without the need to press send. All good result.
When I dial 9-555-5 and wait, nothing happens
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
process_rfc3389: Comfort noise support incomplete in Asterisk (RFC
3389). Please turn off on client if possible. Client IP: xx.xxx.xxx.xx
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dr. Michael J.
Chudobiak
Sent: September 7, 2006 1:39 PM
To: Asterisk
her problems).
Did I misread the Asterisk wiki pages, because I believed that when a
pattern was present, the pattern takes precedence over any "real"
extensions? (i.e. if I have both 1234 and _1XXX as extensions in a context)?
Thanks John, I appreciate all the info.
Mike
___
it's capricious, but we have the users we
have...
Yes, I have read the
admin manual, but couldn't find the info. I am assuming I just don't know
what to look for, but that this functionality exists.
Mike
___
--Bandwidth and
It certainly makes sense, and I tried it...it works, you are right.
So what do you make of this page :
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf
+sorting
Mike
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
ind the led
pattern referring the message waiting in the sip.cfg file. Is this at all
possible?
Mike
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t;context_a" to the CLI
Is it possible, and
if so what is the name of the variable I should use?
Mike
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I'm trying to make
the Polycom 501 go off-hook (in speaker phone mode) when any digits is dialed
and the handset hasnt been lifted. Is this
possible?
Mike
___
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asterisk-users mailing
Dear list
which hardware solution would you suggest for connecting 60 analog
phones to asterisk ?
thank you very much
.mike
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> Maybe a Top-Gate SIP gateway. It supports 16, 24 or 48 FXS ports.
> You don't need T1 or E1 extra in the Asterisk machine, only one ethernet card.
thanks very much to everyone for the comments and the suggestions !
___
--Bandwidth and Colocation prov
r a call to somebody else. Ex: Prospect calls extension 201, talks
to the salesgy, who forwards him to the tech guru somehow.
My guess is I have
to use the transfer feature found in feature.conf. I tried, no
success. What Wiki page do I need to look at to get details on
t
Ah. I'd like to know what others think, but if you're right than it's a
lost cause.
I thought Asterisk kept some sort of control over the call.
Mike
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Steve Glaus
>
xecuting NoOp("SIP/11.11.11.11", "") in new
stack
I am running
1.2.4. Not even sure what the warning means (WARNING[24892]:
app_addon_sql_mysql.c:115 find_identifier: Identifier 2, identifier_type 2 not
found in identifier list).
Any help is
appreciated.
Hi,
Where should I go
to get the Polycom`s latest official (non-beta) version? I am
registered on the Polycom customer website but that doesn't seem
accessible.
Regards,
Mike
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support virtual users (i.e. users not necessarily Linux
users)
3) Can I make this
work with a self-signed certificate? If so, anything in particular that I need
to know?
Regards,
Mike
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asterisk
.
Does any one have
had the same problem and found what the problem was?
Mike
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Im replying to me own message to avoid having somebody
write a lengthy response for nothingturns out my problem is Pure-FTP that's
for some reason not letting the file go through properly.
I'm therefore taking my discussion over to some other
mailing list. Sorry about th
know I am not wasting my time.
PS: If there is a
better FTP server suggestion Ill take it, but one of my "must-haves" is easy of
use and virtual users functionality (with different chroot
folders).
Mike
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running
bootrom 3.2.2 and SIP 2.0.1 (fairly recent).
Mike
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nce at the beginning of my calls. My Asterisk
server is not behind a NAT, so in theory it should work flawlessly. Also,
the latency between my LAN and my Asterisk server is about 10ms, very
stable.
I am trying to figure it out with Ethereal (first thing I
did) but I'm not sure what to l
p.ld file
where I had to, but the phone wont pick it up.
Short of that, can
somebody point me to the newest firmware (2.0.2) to see if that would
help?
Mike
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Any hints on downgrading? I placed the old SIP 1.6.7
on the right folder, but my phone wont pick it up and install it. It must
be thinking "this is an old version, ignore" or
something
I`ve never downgraded a phone, I tend to like upgrading
more :-)
Mike
Fr
remotely, and I did for a few phones, but my
paranoid self would like to double check and see if the sip.ld 1.6.7
re-installed ok by checking the current version. Is that even
possible?
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rick
SmithSent: November
until the end
of a call of the phone is being used.
Mike, happy to contribute answers instead of questions for
once.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rick
SmithSent: November 7, 2006 8:44 PMTo: 'Asterisk Users
Mailing List - Non-Commerci
416123 |
89 | Dial |
My question is, if the caller spends 28 seconds listening to options before
dialing an extension, and the call last 89 seconds...Should the first leg
have a billsec of 89+28=117sec and the second 89 seconds?
Mike
__
I`m impressed. Thanks for the reply, I'll try that!
Mike
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Benny Amorsen
> Sent: November 21, 2006 4:17 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-
Just curious if anyone knows of any hacks to enable announce entry/exit
in MeetMe conferences with SIP (as opposed to ZAP) channels since the |i
option will not work with SIP.
Thanks,
Mike
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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.digiu
;s say I had 50
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
5-555-1234").
At first sight, the two values must be identical. Is there any way to
change that?
Mike
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and voilà!
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
___
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
has 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 '
I haven't changed my configuration in ages. What could be the cause of this
suddent appearan
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
es on the 501 is not
necessarily 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 33
I know ATAs are mostly used 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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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:
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 Me
mple
(i.e. "end call") need to be pressed 2-3 times 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 probl
Somebody was helpful enough to give me the very latest release of Polycom's
firmware (2.1.0). Unfortunately, I still get that issue.
So I'm stuck asking again: Anybody ever got that?
Mike
_
From: Mike [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 11, 2007 13:37
To:
Exactly. It's a weird issue, and I can't imagine what the problem is,
except maybe for bad phones (but then again, why would the phones be only
bad with 2.x?)
UnlessI have bootrom 3.2.2.0019. Is that what people running thelatest
have?
Mike
-Original Message-
Fr
he pre-2.x phone is running with CPU load approaching 0% (0%-7%). The 2.x
phone has tons of spikes in the 100% range.
What could be causing this? Where do I start looking?
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Fuermann
Sent: Thursday, April 12, 200
from then? Are there any consequences?
Using 1.2.13.
Mike
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alashov
Sent: Thursday, April 12, 2007 23:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Huh? IP address ending with 611
Can you do a packet capture and see what the actual contact (Via) in fact
says right before it hits Asterisk?
On Thu, 12 Apr
Nah, nothing of the sort. It's actually a phone using Dynamic IP (so I
didn't chose his IP) and that weird address seems like the NAT device's
address (since I have two phones at the same location.)
Now, how the NAT device over there ended up with this address is
pro
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