: Re: [asterisk-users] Setting custom field in CDR
>
> Mike wrote:
> > Hi,
> >
> > The Asterisk Wiki (page:
> > http://www.voip-info.org/wiki/view/Asterisk+func+cdr)
> mentions I can
> > set any custom CDR field I want. Here is the example it gives:
Hi,
The Asterisk Wiki (page:
http://www.voip-info.org/wiki/view/Asterisk+func+cdr) mentions I can set any
custom CDR field I want. Here is the example it gives:
; Update our accountcode field and then save some random music facts too
exten => s,1,Set(CDR(accountcode)=8675309)
exten => s,2,Se
dtime/localtime.c -> stdtime/localtime.o
stdtime/localtime.c: In function `localsub':
stdtime/localtime.c:1136: error: structure has no member named `tm_gmtoff'
gmake[1]: *** [stdtime/localtime.o] Error 1
gmake: *** [main] Error 2
Thanks,
Mike Clark
_
Hi,
Is there a way to have a Do-While sort of loop, as opposed to a simple
While?
I have a condition that the loop depends on even for the first iteration, as
it often happens in life.
Regards,
Mike
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fline for 60 minutes, than the
queue can try him again.
Is this (or something similar) possible?
Mike
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Then I could just make "downstream-phones" my current outbound context and
everything would do what I'm after. I got what you're saying.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Dave Mille
*bump*
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 20, 2007 12:27 PM
Subject: [asterisk-users] e911
One of my providers has a different SIP
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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"asterisk". Which is NOT what I
want.
Is there a standard way to say "hid my number"?
Mike
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?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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Hi,just spotted this on the RIM site.
http://na.blackberry.com/eng/services/blackberry_mvs/
Just wondered if there is anybody working on somehow linking MVS and
Asterisk, or if it is even possible?
thanks
Mike
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this problem?
Thanks
Mike
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Well, my Zoiper works just fine with Asterisk 1.4.13 and this in the
iax.conf:
[fred]
type=friend
username=fred
secret=abc123
host=dynamic
context=default
NB. I have [fred] and username=fred
What does your configuration look like?
Mike
sean darcy wrote:
> On 10/30
s context?
3. why/how did it just fix itself while I was testing?
4. how come the box receiving the call (gate.tubby.org) answers the call
request with a Reject but doesn't display any debug?
My guess/hunch is that there is some sort of multibyte coding (UTF-8?)
problem with some result codes, s
Michiel van Baak wrote:
> On 18:51, Tue 23 Oct 07, WipeOut wrote:
>
>> Anyone had any experience with an Asterisk server as a VMWare virtual
>> machine?
>>
>
> We are running multiple sites as a VMWare virtual machine.
> All of them are voip only, so I have no idea how it works
> with T1/E
1.2.18?? Care to share how you
compiled it.
Many thanks
Mike
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>
> I've deployed several setups internally using X-Lite and these headsets:
>
> http://www.newegg.com/Product/Product.aspx?Item=N82E16826275009
>
> Haven't heard of a single problem thus far.
>
> -erik
>
Erik:
Do they play well with Vista?
Mike Clark
_
shadowym wrote:
> Or your could use a touch screen with Flash Operator Panel. Just a
> suggestion out of left field.
>
>
shadowym:
Do you have a specific setup w/touchscreen that you have deployed and
that works well?
T
k knows how to handle G729 calls.
>
> Where do I go from here???
>
> Thanks,
> Scott
>
> ___
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gt; Please contact me if you would like to participate or just be in the loop.
>
> Thanks.
>
> JR
>
I certainly want to be in the loop and will help any way I can.
Mike Clark
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;rport
From: "asterisk" ;tag=as7d7e63d3
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
Content-Length: 0
|
10/05/2007 15:30:17.381 CCM|EnvProcessUdpPort -
EnvProcessUdpHandler::fireSignal() varId =
0|
10/05/2007 15:30:17.381 CCM|EnvProcessUdpPort -
EnvProcessUdpHand
sk to transcode and "do the right thing."
What am I missing?
TIA,
--
Mike Diehl
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aster
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
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
On Sunday 23 September 2007 06:43:54 pm Paul wrote:
> Mike Diehl wrote:
> >I just had a user complain about a call getting dropped and another one
> > failing to go through.
> >
> >I'm trying to interpret the log entries for each call and would like to
> >
[EMAIL PROTECTED]
168.0.100 for seqno 102 (Critical Response)
[Sep 23 08:40:21] WARNING[21450] chan_sip.c: Hanging up call [EMAIL PROTECTED]
- no reply to our critical packet.
===
Am I reading and understanding these log
isn't GPL.
http://www.mexuar.com/products_sdk.shtml
Mike Clark
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To
IP addresses. I can
also make outbound calls.
So... apparently Asterisk is working except for the servers aren't showing
up in the peer list.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Erik Anderson" <[EMA
*bump*
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Mike Hammett" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, September 07, 2007 3:25 PM
Su
n,
the Asterisk servers seem to only send RTP packets to the phones private
address.
> Mike Clark wrote:
>
>> Chris Mason (Lists) wrote:
>>
>>> Mike Clark wrote:
>>>
>>>
>>>
>>>> Yes, the Asterisk boxes were
Chris Mason (Lists) wrote:
> Mike Clark wrote:
>
>
>> Yes, the Asterisk boxes were on private addresses. The Polycoms are also
>> behind a NAT. Yes, I tried using externip in sip.conf and this allowed
>> registration, and calls to be placed, but no audio. Unfortun
Jeff Bachtel wrote:
> On Tue, Sep 11, 2007 at 10:32:14AM -0400, Mike Clark wrote:
>
>> We have gotten stuck trying to get a highly available Asterisk cluster
>> fully functional. We used Linux-HA with Asterisk 1.4.11 on privtae IP's
>> behind the virtual publi
lycom
phones.
Thanks,
Mike Clark
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If it has nothing to do with Asterisk, then why does every other device work
as its supposed to?
An MGCP ATA routes out that interface.
A laptop routes out that interface.
That server traceroutes out that interface.
Asterisk doesn't link up.
-
Mike Hammett
Intelligent Computing Solu
*bump*
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent: Thursday, September 06, 2007 10:05 AM
Subject: [asterisk-users] Different Networks
I have multiple
t=outbound-scripted
accountcode=12
callerid=*
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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--Ban
and I appreciate it much.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Jared Smith" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, September 05
*nods* I verified more than once and even copied + pasted to make sure.
Obviously my ping message went through, but my others have not.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: "Bill Andersen" <[EMAIL P
Agreed. This conversation is working just fine, but the important messages
I'm trying to get to go through aren't.
I've never had consistent success from posting to asterisk-users.
Asterisk-biz seems to work all of the time.
-
Mike Hammett
Intelligent Computing Solutions
I've been trying to send messages to the list for the past 24 hours, but they
just aren't going through.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: Mike Hammett
To: asterisk-users@lists.digium.com
Sent:
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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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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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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Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Mike Clark <[EMAIL PROTECTED]> wrote:
>
>> JR Richardson wrote:
>>
>>> I'm interested in putting together a new-user tutorial about DUNDi
>>> configuration and setup. There
me clustering was the only way I would
have gotten it up. Everything else on the wiki is simply too complicated
and not well explained. And it still wasn't a piece of cake, even with
you document. So yes, additional "cookbook"
once you know Asterisk well, you can use a pre-packaged solution
for what it does well, and usually be able to do custom dialplan work in
conjunction and get the best of both worlds.
Mike Clark
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Tzafrir Cohen wrote:
> On Thu, Aug 16, 2007 at 08:30:43AM -0400, Mike Clark wrote:
>
>> Tzafrir Cohen wrote:
>>
>>> On Wed, Aug 15, 2007 at 06:12:09PM -0400, Mike Clark wrote:
>>>
>>>
>>>> Are there any nice GUIs out there
Tzafrir Cohen wrote:
> On Wed, Aug 15, 2007 at 06:12:09PM -0400, Mike Clark wrote:
>
>> Are there any nice GUIs out there for Asterisk Realtime? Google doesn't
>> yield much. I spent a day trying to get VoiceOne to work without much
>> success.
>>
>
Senad Jordanovic wrote:
> Mike Clark wrote:
>
>> Are there any nice GUIs out there for Asterisk Realtime? Google
>> doesn't yield much. I spent a day trying to get VoiceOne to work
>> without much success.
>>
>> Thanks,
>>
>> Mike C
Are there any nice GUIs out there for Asterisk Realtime? Google doesn't
yield much. I spent a day trying to get VoiceOne to work without much
success.
Thanks,
Mike Clark
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mitcheloc wrote:
> Nitesh,
>
> They claim to support numbers on their website so I would say yes.
>
> On 8/11/07, Nitesh Divecha <[EMAIL PROTECTED]> wrote:
>
>> Dean,
>>
>> Can the LumenVox Speech Recognition engine understand numbers?
>> Sorry to ask stupid questions but kinda curious... as fo
? 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
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
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
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-
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
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
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
h Management Group, its subsidiaries, and
> affiliates hereby claim all applicable privileges related to this
> information.
>
> --
> This message has been scanned for viruses and
> dangerous content by *MailScanner* <http://www.mailscanner.info/>, and is
> believed to be clean.
&
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
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.
is subscribe context an addiotional switch/field ?
or its the peer context ?
On 8/9/07, Mike <[EMAIL PROTECTED]> wrote:
>
> I feared so, but I have already started working on this. Thanks for the
> confirmation.
>
> Too bad, the rest of my design was relatively elegan
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> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
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
]: 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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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
or me? Is there some deeper meaning to this
notice I am getting?
Mike
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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
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
(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
n => 12345,hint,Dial(SIP/${A}) ; I need to know ${A} first, but I can't
know before this line is called (it's very DB driven).
What can I do? Am I dead in the water here?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Koh
ne, I need it to all be done in the same Asterisk
priority. See my previous email for background ("Buddy watch and the hint
priority - brain teaser").
Any help is extremely appreciated.
Mike
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exten => _XXX,hint,Set(hint_reg=${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 b
hough all ports 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 si
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
To
al command?
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
akes it do this? Why am I 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 see
John Novack wrote:
>
> Mike Wright wrote:
>
>>Just purchased a Motorola Wildcard X100P ...
>>but the button pressed generates no tone; on button release dialtone returns.
>>
>
> Sure sounds like polarity reversal.
>
Indeed it was. Punch block in th
d fry 1)me, 2)my new FXO
card, 3)my pc, 4)the telco, etc, etc.
Anybody feel up to helping a noobie?
Thanks in advance,
Mike Wright :m)
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my home office and participate weekly on a call
with our corporate office where there are about 10 folks in a conference
room, and I hear them all just fine. A vast improvement over using a
Polycom 501 as a conference room speakerphone
Thanks,
Mike Clark
_
t;
> David Ruggles
> CCNA MCSE (NT) CNA A+
> Network Engineer Safe Data, Inc.
> (910) 285-7200 [EMAIL PROTECTED]
>
We do IVR swith standard dial plan syntax or AEL and do agi calls for
database lookups/transactions. This works well for us.
Mike Clark
___
n the list
(advertising and all) so feel free to email me personally and I'll put you in
touch with them.
Mike Wood
BC Northern Lights
1-866-933-3269 ext 113
1-604-543-1768 (fax)
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan Zakaria
Sent: Wednesda
Haha, I like the full keyboard on my 8700 too much
Maybe when the 8820 is out I will
Mike
On 7/7/07, Dave Bour <[EMAIL PROTECTED]> wrote:
> Time to upgrade ;)
>
> D
> Dave Bour
> Desktop Solution Center
> 905.381.0077
> [EMAIL PROTECTED]
>
> For those who jus
. Shame!
Mike
On 7/7/07, Dave Bour <[EMAIL PROTECTED]> wrote:
> I just tested on mine (7130...non-media supporting yet), in the message,
> it says there's an attachment but the BB itself doesn't register an
> attachment (ie, if on the main email screen, no paperclip on the
On 7/6/07, Dave Bour <[EMAIL PROTECTED]> wrote:
>
>
>
> Can you see an attachment? If so, does it download?
Yes the attachment is there but it seem that it will not download it,
which leads me
to believe it does not understand the format?
thanks,
Mike
> Dave Bour
>
On 7/6/07, C F <[EMAIL PROTECTED]> wrote:
> Have you tried wav49 format?
>
Yes, I have
format=wav49|wav
Mike
> On 7/6/07, Mike Dent <[EMAIL PROTECTED]> wrote:
> > Hi,
> > I recently upgraded the firmware on my Blackberry 8700 to 4.2, this
> > seems to gi
BES 4.1 for sending these emails out via Exchange 2003 if that makes
a difference.
thanks
Mike
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k using the hold button?
Thank you,
Mike Ryan
Installation Support Engineer
Percipia, Inc.
858 Morrison Rd.
Gahanna, OH 43230
+1 614-856-1123 (office)
+1 614-579-6055 (cell)
+1 614-751-2018 (fax)
mykryen (skype)
[EMAIL PROTECTED] (yahoo)
[EMAIL PROTECTED] (msn)
[EMAIL PROTECT
gt; -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mike Diehl
> Sent: Tuesday, June 19, 2007 12:27 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Need to increase call count
>
> Hi all.
>
> I've go
I am looking for a gateway that has several FXS ports and uses IAX. I have
a need for 16 ports, but will accept 6 or 8 port gateways as well.
-
Mike Hammett
Intelligent Computing Solutions
<http://www.ics-il.com> http://www.ics-
.
Any ideas where I can look for improvement?
TIA,
--
Mike Diehl
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Julian
>
>
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Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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org today?
Yes. Me.
Roger.
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asterisk-users mailing list
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Mike
Sales Manager
http://www.v
h and Colocation provided by Easynews.com --
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> > ___
&g
r update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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asterisk
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.80
Now that MCI and Verizon are one, they're probably on legacy MCI. MCI was
also the one that was doing the wholesale SIP pre-merger.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
=> 1234,Rob West
201 => 1234,Julia Zeiter
202 => 1234,Larry Sallberg
This is the phonex.cfg
-
Mike Hammett
Intelligent Computing Solutions
<http://www.ics-il.com> http://www.ics-il.com
___
Why would calls be coming in on the Guest IAX account?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Hammett
Sent: Monday, June 04, 2007 6:56 PM
To: 'Asterisk Users Mailing List
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