Thanks, will take a look. Althought none of those things seem to allow me
to call up my own handler for calls, does it? Or am I misreading?
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, May
=xx,20
Obviously I omitted my cell phone number.
I don't see anything crazy when attaching to asterisk.
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Mike A. Leonetti
As warm as green tea
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system (And more to the
point, allowing easy outgoing routing based on which NIC was used).
Am I correct?
Bonus question if I am indeed correct: how stable is 1.6 right now, compared
to the latest 1.4 (1.4.31)?
Mike
I have 8 aastra phones that are loosing registration. On the phone gui it
says 408 as the registation error after a minute or say they register. In
the cli it eill say the phone is now unreachable then it will show it
registering then available. At first they did it every hour all the phones.
Do you have UDP 1 to 2 port forward to your router?
What kind of router is it?
Respectfully
Michael D Mosier
Ftoc Certified
On May 16, 2010 12:27 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
I have a big problems on my Asterisk systems :
I have one Asterisk Server with static IP
Is it possible to have Asterisk resend the SIP credentials in every INVITE?
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Trixbox ce callcenter works ok but call management is worthless. HUD and the
control panel are not made for call centers. If they ever get hud 3 working
for ce it is suppose to have more call center stuff. Your going to have to
look at 3rd party apps
And yes trixbox questions need to be asked in
This is probably better asked in the asterisknow list.
Log into the console
Type ifconfig to get ip
Go to web browser and type in the ip like
Http://xxx.xxx.xxx.xxx/admin
I forgot the admin passwor google around for it.
Also please read more befor posting there are plenty of sites that eould
I think Asterisk will detect the dtmf for you and the speach recognition
will detect speach.
Respectfully
Michael D Mosier
Ftoc Certified
On May 10, 2010 9:24 PM, Richard Kenner ken...@gnat.com wrote:
On Mon, May 10, 2010 at 7:19 PM, Richard Kenner ken...@gnat.com wrote:
Which speed
)
and the Macro
[macro-broadsmart]
exten = s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: Forbidden from 'Mike A.
Leonetti sip:{broadsmart_us...@broadsmart.net;tag=as6376d669'
The people on the other end sent
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk
On 05/07/10 12:40, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk
that the Asterisk registration doesn't have an
Allow-Events and an Allow in the header. Would this cause any
problems and can this be set in Asterisk to send those in the header?
Thanks.
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Mike A. Leonetti
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wow thanks guys. Ill try it out.
Respectfully
Michael D Mosier
Ftoc Certified
On May 4, 2010 1:36 AM, ad...@3a.hu wrote:
Hello Mike,
On 05-04-2010 06:18, mike mosier wrote:
When DID 713xxx is dialed send an email to mmos...@x...
something like this?
exten = _713X.,1,System(/web/html
whats censored UIN? [VoIP]
On Tue, May 4, 2010 at 8:00 AM, mike mosier trixbo...@gmail.com wrote:
wow thanks guys. Ill try it out.
Respectfully
Michael D Mosier
Ftoc Certified
On May 4, 2010 1:36 AM, ad...@3a.hu wrote:
Hello Mike,
On 05-04-2010 06:18, mike mosier wrote:
When DID
Hey all.
My boss asked me to implement the following
When DID 713xxx is dialed send an email to mmos...@xxx.com. with the
time date and CID included in the email. I know how to code some but am
looking for the best way to do this.
Sorry I might have asked this a couple months back. I
Hey all
What VoIP networking monitoring, asterisk monitoring tools would you
recommend? I started working with an IT company that insists on using DSL
with a Sonicwall router. The problem is that the clients are having sound
problems. The owner is convinced that it's the Asterisk box. In the 4
Howdy all
1. does anyone know a good voip / sip / qos monitoring tool?
2. Has anyone had luck running asterisk phone systems over DSL?
3, Has anyone used sonic wall routers for qos over dsl.
The company I am consulting for would like to install asterisk boxes over
dsl with sonicwall routers.
?
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Ever since the upgrade to Asterisk 1.6 the e-mails from Asterisk are
coming from r...@.
In the voicemail.conf I have
fromstring=Asterisk PBX
serveremail=asterisk
And in my ssmtp.conf
root=asterisk
However they still come from root@
--
?
Otherwise, is there a product/service they can buy that will allow them to fax
to/from their computers?
TIA,
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On Monday 29 March 2010 10:15:50 am jon pounder wrote:
Mike Diehl wrote:
Hi all,
I've cross-posted this to the -users and -biz groups. Hope that's OK.
I have a customer who REALLY needs to be able to send/receive faxes
reliably. I could probably get hylafax configured, but I'm
{
hardware ethernet 00:04:F2:27:8F:F8;
}
host 0004f22afafd{
hardware ethernet 00:04:F2:2A:5A:FD;
}
}
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Hi All,
Anyone one info of where I can get a 'free' DID number ?
I have setup my asterisk box (home) and want to learn more but I need a #.
thanks in advance,
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done.
Let me know if you need more help.
I owe you one, btw, because I read your blog on getting these beasts
provisioned in the first place. sip:15058228...@robodial12.diehlnet.com
--
Take care and have fun,
Mike Diehl
Ok, I see there's alot out there of voip providers.
Curious what to watch out for ? charges and fee's, etc ?
If anyone has feedback as to a GOOD voip provider experience (one that
gave FREE DID) Please share.
Again, I am doing this to learn about asterisk, I'm currently testing
it at home.
My bad, I'm in Los angeles california usa
On Thu, Mar 18, 2010 at 1:06 AM, SIP s...@arcdiv.com wrote:
What country are you in? Makes somewhat of a difference.
N.
On 3/17/2010 8:49 PM, Mike wrote:
Ok, I see there's alot out there of voip providers.
Curious what to watch out for ? charges
Hi Bob,
Thanks for replying. I've thought of doing that, but softkeys are limited
and for a phone with many call appearances (4-5) that would be using many of
the softkeys.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun
?
Mike
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asterisk-users mailing
Michael Leonetti wrote:
I have an a bunch of SPA941 Linksys phones for users in and out of the
office. When the phones are in the office (and on the same network as
the asterisk server) the WMI goes on when it should and off when it
should. But when the phone is on another network and
/madplay -Q -z ---mono -R 8000 -o raw:- -r
-a -12
Format: slin
I have confirmed that madplay is in /usr/bin/. I've also used the
playback command to ensure that the .wav files in the moh0 directory can
be played by Asterisk.
What am I missing?
TIA,
Mike
David Backeberg wrote:
On Mon, Feb 22, 2010 at 2:20 PM, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I'm trying to get moh working on * version 1.4.4. I've setup a test
I don't know the answer, but are you really using 1.4.4? If so,
consider taking some time to review the security
David Backeberg wrote:
On Fri, Feb 19, 2010 at 11:42 AM, Mike A. Leonetti
mleone...@evolutionce.com wrote:
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know
Sean Brady wrote:
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work as expected. Is it possible to then dedicate a physical device
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work as expected. Is it possible to then dedicate a physical device
like a USB port or
this happen once, but I've been unable to reproduce it reliably.
Any ideas?
Mike Diehl.
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A little bit of a strange request.
Basically I want all calls that go to one user go to voicemail
immediately if the user is on the phone. The user is using the Linsys
SPA941, and even though he can be on the phone, calls will still ring
his phone. I tried disabling the rest of the lines on the
Perfect. Thanks.
Mike A. Leonetti
As warm as green tea
Evolution CE
3468C Lawson Boulevard
Oceanside, NY 11572
www.evolutionce.com
516-536-5006 ext 105
516-208-4679 (Direct)
Danny Nicholas wrote:
Set his call-limit to 1 in users.conf. Other than that, you could check the
channel before
I may be late to this thread, but my own restarted every 3-5 days until I
upgraded to 1.4.29 (I skipped 1.4.28).
It`s been running for 8 days now, which isn't long enough for me to declare
whatever-it-is fixed, but enough to say it's at least better with 1.4.29
stability wise.
Mike
it into trunk. Heck, I'll give 200$ for someone just to
tell me how to configure it properly if it's a matter of just missing a
config line.
Mike
Which polycom phones are you using and what SIP firmware are you using?
I am using 3.2.0, with a variety of phones (321, 331, 430, 450, 550
be done as long as the
feature makes it into trunk. Heck, I'll give 200$ for someone just to tell me
how to configure it properly if it's a matter of just missing a config line.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
Hey Jimmy,
3.2.0 is what I am using.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Godbout
Sent: Thursday, February 04, 2010 22:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
.
Where could be the difference? Both are using the same context to dial out.
Mike
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Hi,
I know having Asterisk aware of Polycom Do No Disturb state wasn't working
before (1.4), but is this working in any recent version? Is there any
custom way of doing this?
Regards,
Mike
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Mike Diehl.
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before? Any clues as to how to fix it?
TIA,
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configuration for dnsmasq?
If I can't get this working, I'll have to resort to hard-coding the
information into each of 12 phones Yuck!
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Mike Diehl.
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/asterisk/sounds/ogm/Hold-Ran-4
strategy=rrmemory
joinempty=yes
retry=5
announce-holdtime=no
wrapuptime=15
announce-frequency=60
timeout=10
musicclass=default
autofill=no
ringinuse=no
Any ideas?
Thanks,
Mike Clark
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I find that Siphone works great on the iTouch. Tried it with my own
asterisk box as well as Callcentric and MagicJack and it was very
clear and stable. Haven't played with it since the last firmware
update though as the update removed support for 3rd party headsets .
On 12/14/09, Alex Balashov
Pat Fleet, the original voice of ATT recorded a free set of the
prompts included in Asterisk and also does custom IVR prompts through
her website at http://patfleet.com/ I'm not sure what the going rates
for IVR prompts is, but she charges $15/phrase.
On 12/14/09, Barry L. Kline
Hi Hin,
thanks for the reply back.
Is there a ready-to-go appliance running Elastix? or what type of
hardware do I need to have features as switchvox 305 (for example: it
can handle upto 150 users)
-mike
On Thu, Dec 10, 2009 at 5:03 PM, hin lee hi...@yahoo.com wrote:
I had once considered
call. Can this be done?
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I have Asterisj 1.4.26.`1, and Dahdi 2.2.0.2.
I restarted for no good reason (I was playing around), but it did worry me
that if Dahdi crashed while Asterisk was running that not only Dahdi and
Asterisk would crash, but the whole machine too.
Mike
-Original Message-
From: asterisk
I am new to the list and wanted to get the professionals here input on
Switchvox 305 Appliance ?
List price is 4k, ouch! Is there a better cost-effective way ?
Also feedback (neg/pos) about this appliance.
-mike
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? Ideally, have two values,
one for each T1.
dahdi show channels doesn't show outgoing calls. Is there another command I
am not aware of?
Mike
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Thanks Tim and Danny. It seems a more direct way should be there, but that`ll
work.
Regards,
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Tuesday, December 08, 2009 16:45
To: Asterisk Users Mailing
, but not the whole server!
Mike
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by it being indirect.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Tuesday, December 08, 2009 19:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
Hi,
Running 1.4.26.1 here. I have installed TE420B card in my server, and
followed the appropriate steps (as far as I know to configure it). This
TE420B is connected to a CLEC (T1s), so I am using pri_cpe as singalling
type.
When I dial out, I get this message:
Dec 4 11:37:31]
Forget it, found my issues. I have been looking for hours, but as soon as I
write this I find it. dahdi-channels.conf wasn't included in
chan_dahdi.conf.
That being said, I have other issues now, but at least that one is fixed.
Regards,
Mike
From: asterisk-users-boun
channels correctly,
but not my outbound. My outbound never show up, even during a conversation.
Thanks for helping me figure this out.
Mike
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(although I could
work it out from the former if it was available)
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, December 04, 2009 14:22
To: 'Asterisk Users Mailing List - Non-Commercial
Thank you, at least I am getting the same thing.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Friday, December 04, 2009 16:37
To: Asterisk Users Mailing List - Non-Commercial
/diehlnet.txt
http://www.diehlnet.com/Polycom-0004f211d1d0.txt
I've changed the extensions on the website from .cfg to .txt so that it will
open better for you.
What have I done wrong?
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On Saturday 28 November 2009 06:48:01 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am
as I can.
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On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get it back.
It goes on hold just fine. But when I press the resume button, nothing
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on hold and get
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
The phone is a Polycom 501; it's been discontinued. I am working on a
testing/migration plan to move to the latest Asterisk 1.6.x, but I'm
hesitant to upgrade a system that doesn't currently work right
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working
to fix it?
TIA,
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?
Mike
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On Wed, Nov 25, 2009 at 12:26:10PM +0200, Tzafrir Cohen wrote:
On Tue, Nov 24, 2009 at 11:03:16PM +, Mike wrote:
Folks,
I've got one of those GPO 1950's rotary dial phones that I'm trying to
get working in the UK. I've got pretty much everything working with my
TDM400, the phone
?
Thanks,
Mike.
signature.asc
Description: Digital signature
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plugged the phone directly into the phone line and the dialer
works just fine. Plug it into the TDM400 and it doesn't work, although
I can tap the number usin the hook.
Mike.
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)} is NULL fo the rest of the
dialplan.
My dialplan logic depends heavily on knowing the accountcode.
Any idea what I am missing? Things work well with a normal non-blind
transfer.
Mike
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Just to follow-up: I know there is a variable ${BLINDTRANSFER}. I`d like to
get the CDR out of that channel, but can`t find a way how.
The CHANNEL func gets the info of the current CHANNEL, is there a function
to get variables from another CHANNEL, references by ${BLINDTRANSFER}?
Mike
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On 10/15/09, as asd sa11...@yahoo.com wrote:
plz do not send for me e-mail
thanks
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something?
Thanks in advance,
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
I've tried fourvariations on this theme:
Channel: local/15...@default
Channel: local/15...@dialout
Channel: local/1/default
Channel: local/1/Dialout
Neither one worked. I appreciate your time. Any other ideas?
Mike.
P.S. I thought that setting the context
IP/domain setting in sip.conf to reflect my public IP but it still doesnt
want to work. Thanks to anyone hthat can help me.
-Mike
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, Steve Howes st...@geekinter.net wrote:
On 1 Oct 2009, at 10:43, Mike Bessette wrote:
Hello. I set up an Asterisk box a couple days ago and was having
problems with not being able to hear SIP clients. After some
troubleshooting we have determined that hte INVITE is sending my
local(192.168
OK so basically just uncomment the the localnet settings hten?
On Thu, Oct 1, 2009 at 8:15 AM, Scott L. Lykens slyk...@verimedservices.com
wrote:
Mike –
It looks like you have externip set but no localnet setting.
You need to set localnet for your internal networks so that Asterisk knows
Still no luck. I'm almost ready to start over with a fresh sip.conf and
extensions.conf. Does anyone kno where I can find one without all the
comments and other fluff?
On Thu, Oct 1, 2009 at 9:22 AM, Scott L. Lykens slyk...@verimedservices.com
wrote:
Mike -
Uncomment and set appropriately
Right now I have all firewalls and such turned off. When I have the firewall
enabled, I use the one built in to the Tomato firmware on my Asus router.
How could I determine if this is a PIX/ASA firewall?
On Thu, Oct 1, 2009 at 10:33 AM, Scott L. Lykens
slyk...@verimedservices.com wrote:
Mike
I am looking to configure the asterisk voicemail system to stop asking for
the folder (work, personal, etc) in which to save messages when I do
save them.
Is there any configuration to do this?
Mike
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appreciate any tips.
Mike
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Hi,
yes I did, I did have errors at first but that hurdle has been cleared.
Thanks for the try :-)
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, September 24, 2009
I've tried turning logging way up for the relevant portions of the sip
application, but no telnet. Not sure how I would go about this to get more
info that what I already have. The phone is giving me a response, it's just
that the response
is push message cannot be displayed
Mike
(no calls, lots of
registrations of course, but nothing worth 2Mbits/s)
Regards,
Mike
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Sorry, that is running 1.4.26.1.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, August 13, 2009 23:22
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Stale auth
Philip A. Prindeville wrote:
Anyone have a chance to test any of the various iPhone SIP apps?
I see there are a few out there, but most of the iTunes reviews aren't
sufficiently technical to be useful.
Thanks.
I got iPico and it is working pretty well for me so far.
On Wed, Aug 05, 2009 at 02:13:01PM +0800, D Tucny wrote:
The problem is that your mailbox line was below channel=1, as such, it
applied to the next channel, channel=3 not channel=1...
d
Nice one. Thanks for spotting that.
Mike.
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and needs to be hungup */
if (p-mwimonitor_rpas) {
ast_hangup(chan);
return NULL;
}
}
I have set usecallerid=no on both interfaces and globally but I still
cannot get it to stop.
I have failed to turn anything up on Google regarding this.
Does anyone have any suggestions please?
Mike
=no?
Mike.
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that what happens is that the FXO line rings, so Asterisk rings
the FXS phone as per the extensions.conf, this creates a MWI event which
goes to the voicemail system, which then passes a MWI event to the SIP
phone (as per sip.conf)? Or I could just be talk rubbish!
Mike.
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Sebastian wrote:
Have you solved this issue?
When I restart the machines I can't make an outgoing DAHDI call until I get
an incoming call on that same line.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
, dahdi_cfg
#
# Global data
loadzone= uk
defaultzone = uk
fxoks=1
fxsks=3,4
I have tried only bringing up certain channels but that still fails.
Does anyone have any idea what could be wrong?
Mike.
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