I get this warning in the Asterisk CLI once in a while, and it usually
corresponds with a phone not ringing when it should.
Warning in CLI: Inringing for peer [PEER] < 0
What does it mean and what is the likely cause of this?
___
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this
debugging output? I was calling 8159911010. My server is 208.100.1.33.
Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding
insecure settings, but that didn't seem to solve it on this one.
http://pastebin.com/f5151341f
-
Mike Hammett
Intelligent Computing
.
I have also put both on same linux box (5060 for Asterisk , 506X for
SER) when necessary to meet technical challenges on interface with
specific carriers.
..mike..
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aster
Thanks, to you and Mark, for the quick reply. I used to rely on the Wiki
but it seems I shouldn't
Mike
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Tilghman Lesher
> Sent: Tuesday,
on that one
Mike
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rt sip realtime?
It's a shame Asterisk is (was?) developped with mismatches between .conf
functionality and realtime ones.
Regards,
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent:
gger screen makes juggling calls easier.
Mike
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xed in the latest 1.6, but I haven't tried it)...
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Sunday, December 21, 2008 7:40
To: Asterisk Users Mailing List - Non-Commercial Discussio
phones work perfectly with no perceptible lag, and
Asterisk does send call to them.
Why wouldn`t the delay (in ms) show up and why are they shown offline?
Regards,
Mike
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I will definitely try this later todaythanks!
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
"ManxPower" Wieling
Sent: Monday, December 15, 2008 18:02
To: Asterisk Users Mailing
That would help me, but I can't even do that (send all parked calls to
anybody) because of the dynamic park-dial context.
Regards,
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
"ManxPowe
That information is very much appreciated. Thank you.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson
Sent: Monday, December 15, 2008 11:29
To: Asterisk Users Mailing List - Non-Commercial
Just so I'm clear: there is no way to do what I want short of playing with
the underlying code, correct?
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, December 12, 2008 15:31
To: 'Aste
after the 45 second timeout.
As for show application park, this is not helping.
Regards,
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, December 12, 2008 9:26
To: 'Asterisk Users Mai
Thanks, that makes plenty of sense. I thought I could only check if a
phone as busted it's call-limit, but I just tested and it works well.
Thank you!
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
What am I missing?
Regards,
Mike
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is a "missed call" shown on the screen.
I have access to the Polycom phone.cfg file, and obviously to the Asterisk
.conf files. Anything I can do? Can I send a SIP header to say "don`t show
any call data on the screen"?
Mike
___
tween the "executing Park" cmd and the resulting
messages "Parked
"
When the feature works as designed, both match.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, December 11,
d
back.
What ACTUALLY happens is this:
== Parked SIP/0004f215aabb-0b13d668 on 1...@parkedcalls. Will timeout back to
extension [internal-local-only-hamel] s, 1 in 45 seconds
0004f215aabb is the phone that got put on hold.
Any help is needed, I have been looking at my code/sytem for the l
Thank you for the sanity check!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Darnell
Sent: Wednesday, December 10, 2008 22:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 licenses
> So, in short, i
from outside to a G729 enabled phone and
vice versa, I would reach the limit at 30/30, NOT 15/15.
Right?
I am asking because "show g729" was near 15/15 and I started seeing "codec
unknown" messages in my CLI, and I sure am only using g729 for all
registere
I could, but let's say phone B is limited to local calls, I wouldn`t want
the user to be able to transfer to non-local phone numbers.
Can you explain how your idea makes it simpler or better? I might be missing
the point.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[
later.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Tuesday, December 09, 2008 17:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk variable for SIP context
On Tuesday 09
n
that "forward call" I lose the setvar (I get ''). The value is empty (it's
set, albeit differently, in both phone A and phone B sip entry. So I should
be getting something).
What am I missing?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PRO
l question, so you're
solution would have been good.
I guess SIPPEER func is what is best, I`ll go and see if it works as I think
it does.
Regards,
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Tuesday, December 09, 2008 16:
Great, just what I needed. Thanks!
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton
Sent: Tuesday, December 09, 2008 15:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk variable for
Hi,
Say I wanted to know what context a SIP registration is using to dial out in
my dialplan, what would I do?
For example, I have phones on a "local-calls-only" context (as defined in
sip.conf), others in "unrestricted-calls". In my dialplan, I`d like to act
on that
Thanks, that`s pretty close to what I want. I got confused between members
and agents.
I have enough to go on with this!
Regards,
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Sunday, December 07, 2008 12:54
To: Asterisk Users
queue.conf.
Where exactly do I configure those SIP phones to be part of the queue? Is
something as simple as agent => 1001,SIP/reg_1001 what I need? (or similar?)
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David fire
Sent: Sunday, December 07, 2008 11
December 2008 17:05:28 Mike wrote:
> Thanks, for some reason I had completely missed that cmd.
Verbose was added in 1.2. New to 1.4 was the Log command, designed
specifically for exceptional dialplan conditions where you might wish to
log a set of output for later perusal (like if it happens
eue a queue that calls back
the customer? There is conflicting info when searching for "callback queue".
Mike
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I am almost ashamed not to have thought of it
.
Thanks a lot, that will do perfectly.
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin
Sent: Friday, December 05, 2008 17:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Thanks, for some reason I had completely missed that cmd.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins
Sent: Friday, December 05, 2008 17:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
Hi, all
I want to add more than 200 sip accounts into sip.conf, username from 6000
to 6199, password is the same, i remember there is a better way to do this
case, however, i have not searched the method yet.
Anybody can tell me this method, TIA.
BR
Mike Li
Is there a way, for debugging purpose, to have a level where only Noop()
cmds are shown in the CLI but nothing else in the dialplan appears (except
for errors and warnings or course)?
Mike
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H, not sure about you but I often pick up a book and flick from
the back to the front, does nobody else do that?
On 05/12/2008, Atis Lezdins <[EMAIL PROTECTED]> wrote:
> On Fri, Dec 5, 2008 at 3:47 PM, Apostolos Pantsiopoulos <[EMAIL PROTECTED]>
> wrote:
>>
>>
>> Tzafrir Cohen wrote:
>>
>> Top
et put on hold and need to keep
the call going. If I hang up, so does the caller.
Can`t the parked call just go park itself (and hang up my leg of the call),
and ideally call me back if not picked up within x seconds?
Mike
___
-
found, would be appreciated.
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David fire
Sent: Tuesday, December 02, 2008 19:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dahdi and ztdummy
ok
dont pay attention to that file
card you have a problem in the config.
David
2008/12/2 Mike <[EMAIL PROTECTED]>
Thanks Joseph. I went and read thos pages, nothing helps me. As mentionned
in my other post, I don`t have a /dev/dadhi fileI don`t know why it
wasn`t created or where to go from here.
Mike
-Or
Thanks Joseph. I went and read thos pages, nothing helps me. As mentionned
in my other post, I don`t have a /dev/dadhi fileI don`t know why it
wasn`t created or where to go from here.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph L
line 0: Unable to open master device '/dev/dahdi/ctl
Well that probably explains it, because there is no such file. But as I am
not a linux expert (comfortable linux user at best), I am not sur where to
go next.
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Beha
Asterisks the same way. Where should I be
looking?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Michelson
Sent: Tuesday, December 02, 2008 17:17
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dahdi a
still need the ztdummy timer (or whatever
it`s called now). How do I get myself going?
Regards,
Mike
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Hi,
Is there a way to page a Polycom phone that is already in use (if, of
course, the call isn't on speakerphone already)?
Mike
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asterisk-users mailing li
statement about
having changed nothing.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: Thursday, November 27, 2008 13:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hints stopped
ough to block the would-be hacker
from tyring extensions?
Any help is appreciate, I clearly don't understand SIP peers.
Mike
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something numerical, and not NULL).
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Wednesday, November 26, 2008 11:21
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Hints stopped working suddently
Yes I did
Yes I did. Nothing changes, really. And it all looks good.
What I don't get is why the status "unavailable" appears when the phone is
disconnected, but the status "inuse" doesn't when on a call. That
"unavailable" works fine is some sort of proof tha
Good theory, but I had already tried that (and my phone re-subscribes every
60 seconds anyways)
so that's not it.
Regards,
Mike
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen
Sent: Wednesday, November 26, 2008 10:24
To: Asterisk Users Mailing
Do you use the Asterisk GUI? Changes from it can mess with contexts in the
dialplan (extensions.conf) and the hints need to remain in the [internal]
context.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Wednesday, November 26, 2008 6:33 AM
To: 'Asteris
w hints command) when I dial out or get a call.
Using Asterisk 1.4.22 and Polycom 3.1.1 firmware (latest). And I restarted
asterisk just in case, no help.
Regards,
Mike
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I tried using this iptables sample, and did not see duplicate packets
on '--to-ports' port
Has some verified this is working for them?
I listened on both ports with tcpdump command.
thanks,
On Tue, Nov 18, 2008 at 9:35 PM, Matthew J. Roth <[EMAIL PROTECTED]> wrote:
> Rizwan Hisham wrote:
>> Is
onfigure *everything*.
Mike
On Mon, Nov 17, 2008 at 11:44:33PM -0700, Jesse Molina wrote:
>
> Digging up an old issue here, so please disregard. I'm making this
> statement for historical and searches.
>
> I own a couple of Linksys SRW series switches. The modern/updated
&
the direction of a seller.
I am in the US.
thanks,
Mike
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bit more elegant.
Thanks,
Mike Clark
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e, the UK ringer! Why didn't I think of that? Doh!
Thanks Gordon, I'll order a pair.
Mike.
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anything I can do to make my other phones ring?
Thanks,
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.
>
hanks in advance,
Mike
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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
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
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.
_
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
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
machine.
..mike..
At 11:23 AM 9/28/2008, Gordon Henderson wrote:
>On Sun, 28 Sep 2008, Jim Boykin wrote:
>
> > We plan to use asterisk for conferencing. As I understand, it requires
> > either a separate hardware like x100p clone or ztdummy. What are the
> > pro & cons of
Hey hey...
I'd like to create a new feature code in asterisk so when a user dials...
say... *00, it would then call some other extensions and play a sound file
to them.
So far, this is what I have...
[testing-custom]
exten => *00,1,Wait(1)
exten => *00,2,Playback(beep)
exten => *00,3,Playback(be
See replies in text below:
At 07:09 AM 8/31/2008, Sriram wrote:
Hi
My Scenario is to implement Asterisk in a Call center.. I;ve TE420
Digium card and plan to terminate 4 PRIs (E1) on it. I;ve 30 Agents
inside..Since its a PRI i m not using any hardware echo cancellation
module.The calls would
|row|
context = row[0]
print "\n"
print '[' + context + "]\n"
print "Switch => Realtime/" + context + "\n"
end
Thanks,
Mike
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Jay R. Ashworth wrote:
> On Wed, Aug 13, 2008 at 07:32:09PM -0400, Mike Clark wrote:
>
>> It manages the MySQL database tables directly via RoR. It is
>> specifically built for using Realtime Asterisk. It uses #exec statements
>> in the extensions.conf file to set up
Brandon:
It manages the MySQL database tables directly via RoR. It is
specifically built for using Realtime Asterisk. It uses #exec statements
in the extensions.conf file to set up the context declarations.
Thanks,
Mike
bkruse wrote:
> That's pretty cool, I love ruby. What method doe
,
contributions, and criticisms are welcomed!
Here are the links:
Sourceforge: http://sourceforge.net/projects/ragui/
Website: http://www.ragui.net
Enjoy!
Mike Clark
WebPoint IT Solutions
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THis machine is well OVER POWERED for the task you define.
The greater issue will be the media bandwidth.
I have operated with 220 simultaneous g711 participants
in 18 ~ 20 different conferences.
At 02:45 PM 8/11/2008, you wrote:
Hi list
I got one HP ProLiant DL380 G5 - Quad-Core Xeon E5440 2
; Voicemail for customer '[EMAIL PROTECTED]'
[cust1]
100 => 1234,Mike Oliveras,,,tz=pacific|[EMAIL PROTECTED]
200 => 1234,Grand Stream,,,tz=pacific|[EMAIL PROTECTED]
300 => ,Joe Blow,,,tz=pacific|[EMAIL PROTECTED]
400 => ,sipp,,,tz=pacific|[EMAIL PROTECTED]
800 => 1234,Mich
On Thursday 24 July 2008 12:58:29 am Steve Edwards wrote:
>
> The "agi debug" command (1.2) would have shown you where you violated the
> "protocol."
Nice to know...
--
Mike Diehl
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worked as expected.
Thank you for your time.
Mike.
On Wednesday 23 July 2008 01:48:14 pm David Van Ginneken wrote:
> Mike Diehl wrote:
> > Hi all,
> >
> > I'm trying to build an IVR using the Perl AGI module at
> > http://search.cpan.org/~jamesgol/asterisk-perl-0.10
($msg, 12, 1);
$result = $agi->get_data($msg, 12, 1);
Finally, I tried to use the get_option() method that was documented in the
module POD file; Perl complains that the method isn't defined:
$result = $agi->get_option($msg, "12345", 1);
So, what am I missing? I
the same, just the D channel as 48)
Primary D-channel: 24
Status: Provisioned, Up, Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 100
faxdetect=incoming
>
> In the past, faxdetect has been known to cause problems.
>
I'll change zaptel.conf first so I only change one thing at a time.
Mike
-
The information contained in this e-mail mes
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 w
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`t
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, June 17, 2008 09:23
To: 'Asterisk Users Mailing List -
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
___
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:
>
> Provi
a problem
> with something else before that step.
>
> On Wed, Jun 11, 2008 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 fil
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
_
y, June 10, 2008 09:34
> 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
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
I'll get right on hunting about RDNIS, thank you VERY much! :)
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ata regarding which mobile phone the call
was originally intended for...
Is this a pure pipe dream? does PRI carry call diversion information?
Thanks
Mike
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Non-Commercial Discussion
> 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
> 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
t's a forward, it doesn't work.
Maybe the problem comes from the fact that it's a variable in my sip
database (I am using realtime SIP entries) and not in the diaplan per say.
My setvar column is this:
internal_callerid=blabla <123>;did=551234
I tried a
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
Anyone have recommendations for wireless headsets that work well with
Polycom phones and Asterisk?
Thanks,
Mike Clark
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To UNSUBSCRIBE or update
several days of advanced negotiation to
agree on the methodology with all concerned. This is a typical
situation when you want to make sure the client knows enough to make
a valid decision.
..mike..
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ar off the
typical product specifications
that nothing published by Digium or anyone else could anticipate
those surprises that
come when you least expect.
..mike..
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asterisk-us
risk platform.
If this is your interest, then drop in
http://voipusersconference.org The context of the discussion
is NON-COMMERCIAL. I have no product or service for sale. I am just
discussing a different approach
to using Asterisk.
..mike..
At 09:42 AM 5/16/2008, randulo wrote:
>http:
Well, I answered my own question by (alot of) trial and error: semi-colum is
used to separate the variables.
Mick
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Monday, May 12, 2008 12:20
To: 'Asterisk Users Mailing List - Non-Commercial Discu
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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To UN
y obvious now looking back :-/
I'm happy to say that the guys from redfone were incredibly helpful
every step of the way, without Jose explanations and tips I would
probably still be scratching my head...
Hope this helps another poor soul in my situation out in the future.
Mike
On Wed, May
e... I am waiting until out of
hours tonight >6pm GMT to test to see if these versions on libpri,
zaptel and asterisk fix the issues; and I will update the list to
reflect either my success or failure :/
Thanks guys
Mike
On 5/4/08, Mike Hardman <[EMAIL PROTECTED]> wrote:
> Ok Guys, I
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