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to be in the loop and will help any way I can.
Mike Clark
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)|CLID::FLCVOIPP1-ClusterNID::10.200.204.10
Mike Coons
Network Telecommunication Services
Fort Lewis College
Information Technology
1000 Rim Dr
Durango, CO 81301
Voice: 970-247-7666
[EMAIL PROTECTED]
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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
and do the right thing.
What am I missing?
TIA,
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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
(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 entries correctly?
Thank you for your help,
--
Mike Diehl
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
confirm my understanding.
The first entry is from
://www.mexuar.com/products_sdk.shtml
Mike Clark
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unmonitored]
That's what happens after I do a iax2 show peers. So apparently calls are
coming in, but showing the peers isn't bringing up any 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
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. Unfortunately,
Polycom does not support STUN
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 on private addresses. The Polycoms are also
behind a NAT. Yes, I tried using externip in sip.conf and this allowed
*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
asterisk-users@lists.digium.com
Sent: Friday, September 07, 2007 3:25 PM
Subject: Re
phones.
Thanks,
Mike Clark
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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 public IP. I got as far as getting phones
*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
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 Solutions
-scripted
accountcode=12
callerid=*
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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Mike Hammett
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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: Wednesday
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
http://www.ics
*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 PROTECTED]
To: Asterisk
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
asterisk-users@lists.digium.com
Sent: Wednesday, September 05, 2007
is
not that I want.
How can I make sure that only the external leg is counted?
Mike
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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 is a lot of great information, setup
guides already but the feedback I
too complicated
and not well explained. And it still wasn't a piece of cake, even with
you document. So yes, additional cookbook type documents that
thoroughly explain things should greatly help adoption.
Mike Clark
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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 Clark
Are you looking for open source or commercial?
Senad
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.
GUI that will allow you to control raw asterisk
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 for Asterisk Realtime? Google doesn't
yield much. I spent a day trying
in
conjunction and get the best of both worlds.
Mike Clark
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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 for my
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
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
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Sales Manager
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Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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 elegant (IMO) and easily to
modify
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
://www.api-digital.com--
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Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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
(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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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
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
, 4)the telco, etc, etc.
Anybody feel up to helping a noobie?
Thanks in advance,
Mike Wright :m)
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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 the basement had tip and ring reversed.
Probably been
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
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dial plan syntax or AEL and do agi calls for
database lookups/transactions. This works well for us.
Mike Clark
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on 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: Wednesday, July 18
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 give
it the ability to play wav files.
I wondered
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
Desktop Solution Center
. 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
envelope). It doesn't give
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 just want it to work...
Giving you complete
BES 4.1 for sending these emails out via Exchange 2003 if that makes
a difference.
thanks
Mike
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,
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 PROTECTED] (gtalk)
http://www.percipia.com
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-il.com
]
[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 got a project where I need to make outbound calls and play a
prerecorded .wav file to the called
can look for improvement?
TIA,
--
Mike Diehl
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/
/reginfo
msg msg.bypassInstantMessage=1
msg.mwi.1.subscribe= msg.mwi.1.callBack=299
msg.mwi.1.callBackMode=contact/
/msg
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com http://www.ics-il.com
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
Sent
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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:
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Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Making it happen
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.
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Making
or update options visit:
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Sales Manager
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Making it happen
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Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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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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 - Non
-users mailing list
To UNSUBSCRIBE or update options visit:
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--
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Sales Manager
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Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Sales Manager
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to that customer were going to the default
context, despite the fact that I explicitly defined the context I wanted the
calls to go to in all entries in iax.conf.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com http://www.ics-il.com
set myself up as a Linksys Partner, and have spent hour(s) on the
phone with them, but it still doesn't work. Ideas?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com http://www.ics-il.com
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Hi,
would it be possible to use Asterisk to record calls only? There would
be an existing PBX and calls come in on a ISDN30 line?
The Asterisk box would need to sit between the incoming ISDN 30
circuit and the existing PBX.
Is this possible?
thanks
Mike
applications in this manner.
Google Asterisk AGI and this should get you started.
Mike Clark
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On Thursday 24 May 2007 06:35, Steve Murphy wrote:
On Wed, 2007-05-23 at 20:51 -0600, Mike Diehl wrote:
Hi all,
I'm having a problem with an asterisk server being unable to call certain
cellphones and answering machines. Anytime the person answers the phone
call, everything works well
fails, my friend does hear the phone ring. BTW, I'm running
Asterisk 1.4.4.
Does anyone know how to fix this?
TIA,
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Yeah, I was trying to have it match the caller ID with what they're dialing
so that I don't have a separate entry for every customer.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
the voicemail system? If they call their own number, how do I get Asterisk
to recognize that and take them to the voicemail system?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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Hi,
is there a way or feature available in Asterisk where one can 'pull' a
call back from
voicemail.
i.e. if you don't get to the phone in time and it goes to voicemail,
can you key some
sequence in and pull the caller out of voicemail and speak to them?
Thanks
Mike
If it is easy, could you enlighten me? I have another thread on caller ID
matching, but I haven't received any positive responses.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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asterisk-users
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Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647
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
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
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
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