On Fri, Jun 29, 2012 at 10:42 PM, Michelle Dupuis mdup...@ocg.ca wrote:
Do the C610H and C300IP use an international standard for frequencies? I
can't even find gigaset sold in USA/Canada...
Gigaset C610a (base + handset combo) are widely available, even on
Best Buy and Amazon. You can add
Anyone have an update as to when Digium will ship a working package?
-- Forwarded message --
From: Andrew Joakimsen joakim...@gmail.com
Date: Wed, Mar 23, 2011 at 23:53
Subject: Issues with Digum Repos / AsteriskNOW Bad Packages
To: Asterisk Users Mailing List - Non-Commercial
asterisk-l...@puzzled.xs4all.nl wrote:
On 06/06/2011 08:07 PM, Andrew Joakimsen wrote:
Anyone have an update as to when Digium will ship a working package?
According to https://issues.asterisk.org/view.php?id=18748 new packages
should already have been pushed. If not perhaps you could join
I am still using Asterisk 1.4 because of the Asterisk GUI. I don't
understand why it was ever dropped, it's easy to setup (no SQL
databases), quick, works well and in my experiance it gets along with
manual config file changes.
The only real issue I've encountered with 1.4 is Digium can't seem to
It isn't any better than the so called t.38 support in Asterisk that
only drops calls. Gee I wonder why, maybe so they can sell their fax
product?
On Wednesday, May 4, 2011, Steve Edwards asterisk@sedwards.com wrote:
On Wed, 4 May 2011, vip killa wrote:
screw that i just got hylafax
On Thu, Mar 24, 2011 at 15:34, Kevin P. Fleming kpflem...@digium.com wrote:
On 03/23/2011 10:53 PM, Andrew Joakimsen wrote:
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap
voicemail storage and Asterisk 1.4.
After having installed AsteriskNOW with Asterisk 1.4
On Thu, Mar 24, 2011 at 15:34, Kevin P. Fleming kpflem...@digium.com wrote:
On 03/23/2011 10:53 PM, Andrew Joakimsen wrote:
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap
voicemail storage and Asterisk 1.4.
After having installed AsteriskNOW with Asterisk 1.4
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap
voicemail storage and Asterisk 1.4.
After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI
I run the yum package manager and replace voicemail with imap
voicemail and attempt to start Asterisk, however the
2011/3/3 Piotr Górski pi...@prnet.pl:
Something free?
If your provider provides a proper rate table you will pretty much
know which is mobile and which is fixed line and assuming their
rates are accurate I assume your company wouldn't care if you allowed
the mobiles billed at fixed line
On Thu, Dec 2, 2010 at 11:58, Jeff LaCoursiere j...@sunfone.com wrote:
we have a low-cost Atom based PBX and a fax relay setup locally with
hylafax/iaxmodem to solve that issue, and it is working very well. We
don't however, have a solution for their alarm lines.
You would desire the entire
On Sat, Feb 12, 2011 at 07:31, ast guy ast...@gmail.com wrote:
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking for SIP hardphone that works well with
asterisk server.
Polycom phones are still working well and durable as a brick.
Has anyone gotten one-touch call parking to work on Polycom phones via
the Enhanced Feature Keys feature working? I've looked at various
examples, they appear correct, but the phones (501, 3.1.x firmware)
show the Park button while in a call but this does not actually cause
the call to be parked.
As recent as 2008 Asterisk 1.4 is feature frozen if that is the case
how come now CallingToken support is added? I don't really know what
this is but all I know is:
1) Callingtoken adds new options to the config files
2) Callingtoken is some new protocol in IAX?
3) Upgrading asterisk 1.4 breaks
I don't see why it does not work. Setting RDNIS and calling most GSM
mobile phones produces a forwarded call annoucement, so why would
the do it any different? We get RDNIS in a SIP field and use it to
keep the same voicemail for a desk phone and cell phone, also can
forward ILEC and most CLEC
I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on
the same machine. After rebooting the iaxmodems don't register to
asterisk. Stoping and starting the relevant services gets it working,
but what is the point of using init scripts if it does not work right?
I already tried to
Does anyone know what version of Asterisk Switchvox uses, and if it is
modified in any way? FWIW, I am dealing with a provider that claims
compatibility with Switchvox but not Asterisk for their SIP trunking
service.
--
_
--
This works for me using DNSMasq:
dhcp-host=00:04:f2:*:*:*,net:polycom # creates a 'polycom' group for all
equipment with MAC prefix of 0004f2
dhcp-range=net:polycom,192.168.1.151,192.168.1.180 # dhcp range for
'polycom' group
dhcp-option=net:polycom,66,http://pbxserver/gui/phoneprov; # polycom
It is a problem with Windows mobile phones as well, there is *NO* way
to dial a number e.g. 800-CALL-ATT. On my Nokia S60 phone (E71) I can
dial the number but it is not possible to dial letters when the call
is connected.
This affects everyone. When I call American Express it asks me to
enter my
for Exchange between WiFi and 3G automatically.
On Thu, May 14, 2009 at 02:02, Remco Barendse aster...@barendse.to wrote:
On Tue, 12 May 2009, Andrew Joakimsen wrote:
Overall, given the limitations of WiFi, it works rather well. I've
never had to reboot my E71 or play with the settings after
On Mon, May 11, 2009 at 11:24, Cory Andrews c...@voipsupply.com wrote:
Anyone using Nokia “E” Series handsets with Asterisk? I’m trying to deploy
some e71’s and am having an issue. I can get a single handset working, but
when I try to create a SIP profile on the second phone, it won’t allow
I use these cards and they work pretty well. FWIW when Digium sold
them they were also just winmodems with a resistor removed to change
the PCI device ID. Later on the Zaptel driver included the device ID
of the winmodem.
I used to be able to get the winmodem itself for under $10, but I
think
On Sat, Apr 25, 2009 at 06:03, sean darcy seandar...@gmail.com wrote:
We're getting a new server. I'm considering installing 64bit fedora
rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any
issues we should expect?
I have been using Asterisk on 64-bit and 32-bit openSUSE
The *BEST* solution would be to have Verizon switch you over to a PRI.
On Wed, Apr 29, 2009 at 17:29, Daniel Hazelbaker
dan...@highdesertchurch.com wrote:
Okay, I can't find what might be causing this. Here is what I got:
Asterisk server hooked up to a digital T1 line (full 24-channel) via a
There are RAID controllers (hardware, of course) that have battery
backup, so the risk in very minimal in using write cache. Just one
(random) example:
http://h18000.www1.hp.com/products/servers/proliantstorage/arraycontrollers/smartarrayp400/index.html
SAS controllers support SAS and SATA
Could you explain better what you want to do? The VeriFone terminal
can talk to the merchant processor via a phone line or via Ethernet
(TCP/IP). Why do you need to interpret the incoming information from
the VeriFone? What do you intended to do with that information?
On Tue, Apr 28, 2009 at
Google shows one result for low cost ATA:
http://www.trixbox.org/forums/vendor-forums-non-certified/linksys-cisco/linksys-pap2-and-rt31p2-low-price
Buyer beware! Those are probably counterfeit!
On Fri, Apr 24, 2009 at 19:15, Wilton Helm wh...@compuserve.com wrote:
Have you checked ebay?
Just
On Mon, Mar 16, 2009 at 20:26, Marc Charbonneau timebandit...@gmail.com wrote:
I was looking at the aastra 9133i, however I was informed that this phone is
no longer supported. What are good phones around the $100 - $125 price
point? (Need POE)
I like the Polycom IP-330. 2 lines, nice
Is it the Windows software, or other? I noticed the Nokia E71 mobile
has an option for Cisco IP Communicator (besides the built-in SIP
client)
On Wed, Mar 4, 2009 at 22:32, Dorien K. Takeshi
dorien.take...@webhad.co.nz wrote:
Hi guys,
Has anyone had any luck with getting the Cisco IP
On Mon, Feb 23, 2009 at 03:10, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Mon, Feb 23, 2009 at 08:00:38AM +, Gordon Henderson wrote:
On Sun, 22 Feb 2009, Doug wrote:
Interesting shopping list - I've just built a new server for my co-lo and
it's an Intel Atom mobo. Normally I do use
On Wed, Feb 18, 2009 at 12:50, bilal ghayyad bilmar...@yahoo.com wrote:
And is there a bank accept to give such kind of communication?
The user was able to dial his card number and the amount from his phone (or
IP Phone registered with Asterisk), and Asterisk communicate with the bank or
On Tue, Feb 17, 2009 at 15:09, Jeff LaCoursiere j...@jeff.net wrote:
On Tue, 17 Feb 2009, Jerry Jones wrote:
Most alarm systems around here use bursts of dtmf - not an actual
modem to communicate with alarm central.
Yes I have seen these have many issues with voip in the path.
You mean
On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere j...@jeff.net wrote:
Anyone have much luck with these on ATA's? I have a few sites that use
them succesfully with multi-port Audiocodes boxes, but just connected ten
machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb
On Thu, Jan 22, 2009 at 19:34, Rehan Allah Wala re...@supertec.com wrote:
Your service is still up and working,
Because Suzanne Bowen has better judgment than you.
You did charge back on the payment to us,
That is correct. There is $86 balance in my account I did not expect
to get back by
.
-- Forwarded message --
From: Rehan Allah Wala re...@supertec.com
Date: Sat, Jan 10, 2009 at 13:56
Subject: Your DIDX account
To: Andrew Joakimsen joakim...@gmail.com
Cc: muneeb @ supertec. com mun...@supertec.com, suza...@supertec.com
Thank You for this email Andrew,
Please move your numbers
On Mon, Jan 12, 2009 at 16:15, Karl Fife karlf...@gmail.com wrote:
QUESTION: Who's in the wrong:
I recently saw an example of a u-law file with a metadata header on the
file.
The asterisk playback function 'PLAYED' the ascii header values as if they
were audio data, creating an audible
On Wed, Jan 7, 2009 at 23:39, Noah Miller noahisaacmil...@gmail.com wrote:
Hi Mark -
You really want to do SLA with all 23 lines of the PRI? That's a
lotta lines to be shared. You'd need two sidecars for each phone
(Cisco or Polycom).
Actually there will be multiple PRI's :)
This
On Wed, Dec 31, 2008 at 22:09, Paul Hales pdha...@optusnet.com.au wrote:
Karl Fife wrote:
Allison Smith just created a hysterical parody music on hold Parody.
Whatever you were doing, stop, and dial this number to listen to it:
360-519-5689. 2 minutes.
I just gave her a few ideas, but she
On Tue, Dec 30, 2008 at 00:25, Jeff LaCoursiere j...@jeff.net wrote:
On Mon, 29 Dec 2008, Andrew Joakimsen wrote:
On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote:
What does Audiocodes release under GPL?
j
The MP-202 is running Linux. At first they said no it's
AudioCodes blatantly violates the terms of the GPL by not distributing
the source code even after requesting it. Please don't use their
hardware.
On Thu, Jul 24, 2008 at 07:34, Frank Tarczynski ft...@mindspring.com wrote:
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It
On Mon, Dec 29, 2008 at 17:25, Jeff LaCoursiere j...@jeff.net wrote:
What does Audiocodes release under GPL?
j
The MP-202 is running Linux. At first they said no it's not and
later they admitted it did, but refused to supply the source code.
Oddly enough, the Linux distribution is OpenRG,
On Mon, Dec 22, 2008 at 16:30, amit salunkhe amitsalunkh...@gmail.com wrote:
i need to implement Inward SIP usring dialing in my Asterisk IPpbx,
So anybody can recah me by dialing my SIP uri. same time my DNS on same
server where currently Asterisk running.
how ican implement this.
On Fri, Dec 19, 2008 at 12:08, David fire ddf...@gmail.com wrote:
set(CALLERID(number)=000)
David
Keep in mind that with doing that, you would loose the caller ID
number for the CDR -- thus there will be no record of the caller ID
anywhere (asterisk-related, at least).
I believe if you use a
I am looking for a VOIP provider that can offer origination and
provide the RDNIS with each call. I am not looking for any large
volume commitment.
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On Fri, Dec 12, 2008 at 18:57, Eric ManxPower Wieling e...@fnords.org wrote:
Philipp Kempgen wrote:
michel freiha schrieb:
I would like to ask please if there is a way to play a ring back tone from
asterisk when the customer try to make a call...I already added the ringing
function to the
On Tue, Nov 18, 2008 at 22:30, mark morreny [EMAIL PROTECTED] wrote:
Hi,
Is it possible to connect Asterisk with a mobile base station to handle call
switching? What kind of protocol will I need to use to convert to sip?
Any pointer or info will be greatly appreciated.
There are various
Anyone else notice all of DIDx is down? Calls on their 3rd-party DIDs
do not go through, but the website is up.
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To UNSUBSCRIBE or update options
On Mon, Nov 17, 2008 at 11:55, Bipin [EMAIL PROTECTED] wrote:
Hello all,
I am facing as serious problem when running asterisk in HP server.We are
developing application to make the outbound calls in PRI lines .We normally
uses IBM machine as our servers ,and it was working fine for all
will be more prone to purchase Polycom if updates are freely
available.
On Mon, Oct 27, 2008 at 07:39, Dave Fullerton
[EMAIL PROTECTED] wrote:
Tilghman Lesher wrote:
On Sunday 26 October 2008 21:28:34 Andrew Joakimsen wrote:
Other vendors, including Cisco, will provide the firmware directly. I
On Sun, Oct 5, 2008 at 8:04 PM, David Backeberg [EMAIL PROTECTED] wrote:
Isn't IMAP IMAP? Does MS not actually follow the protocol? Why would
it be different?
When I setup my voicemail.conf for IMAP Asterisk does not work right.
sip show peers only shows 1 peer. The CLI is freezing up, etc.
On Sun, Oct 26, 2008 at 4:51 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Sat, 25 Oct 2008, Joseph L. Casale wrote:
X100P.
Yeah I saw these but they are single port and I need at least 2 ports. I
only have 1 free pci slot as well.
OpenVox.
Those look great, and on top of the price
On Fri, Oct 24, 2008 at 10:19 PM, Chris Walton [EMAIL PROTECTED] wrote:
The 3.1.0 firmware allows you to create up to 10 custom softkeys.
This is all documented in Polycom's SIP 3.1 Admin Guide.
Should I post some examples?
Which would be great, if Polycom weren't the Firmware-Nazis that they
On Fri, Oct 24, 2008 at 10:09 AM, Drew Gibson [EMAIL PROTECTED] wrote:
Can anyone clarify how SMS to non-mobile numbers are generally handled
in North America?
Is it possible to have SMS delivered direct to your landline DIDs? Then
have Asterisk relay it to the actual mobile DID.
When I send
: Andrew Joakimsen
Sender:
To: Asterisk Users Mailing List - Non-Commercial Discussion
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey
Sent: Oct 26, 2008 5:45 PM
On Fri, Oct 24, 2008 at 10:19 PM, Chris Walton
No, it is not possible. I submitted a bug report[1], because it has
been bothering me too.
[1]http://bugs.digium.com/view.php?id=13781
On Thu, Oct 23, 2008 at 4:36 PM, Mark Wiater [EMAIL PROTECTED] wrote:
Hello all,
I've got dialout= and callback= set in my voicemail.conf so that I
can have
If you are VoIP-only then you need a SIP provider that offers T.38.
On Wed, Oct 22, 2008 at 11:17 PM, Brendan Martens
[EMAIL PROTECTED] wrote:
I am using 1.6.0.1 and we are going to be pure voip. I know it has
pass through and termination, but that is useless if I don't have a
way to transform
Could it be DND? I noticed the other day on my 501 that if I set do
not disturb the phone still rings -- it is just silent. This could
be caused by the configuration, I am unsure.
On Sat, Oct 18, 2008 at 3:16 PM, Bill Michaelson [EMAIL PROTECTED] wrote:
I recently sent this email to a user in
On Sun, Oct 19, 2008 at 12:31 AM, Stephen Reese [EMAIL PROTECTED] wrote:
My latency is kind of high and the voice delay is noticeable.
Then pretty much all you can do is lower the latency to lower the
voice delay, or use a connection to th e PSTN that has a marginally
lower delay if you have no
Does anyone know if I have an older Telrad PBX if I can get CallerID
to Asterisk when the connection is via analog FXO-FXS? I only need 1
or 2 lines so T1 is an overkill.
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I've seen something like that (in your next post you show 20-some
modules, a stock install will be 100 modules) when using the
openSUSE distribution Asterisk package along with asterisk-addons
package. What happens is it gets stuck on some module that is not
configured I think one of the ones
On Sun, Oct 12, 2008 at 5:17 PM, sean darcy [EMAIL PROTECTED] wrote:
Becasue of all the issues with fax over voip, we want to use pstn for
our fax machine, but not dedicate a line just to fax.
I'm thinking of having asterisk answer the pstn line, check for fax
tones, and route appropriately.
Are you using NAT?
On Fri, Oct 10, 2008 at 4:24 PM, Paul Douglas Franklin [EMAIL PROTECTED]
wrote:
We have 3 Grandstream Budge Tone 100 phones which are being very fluid
on incoming calls. They are set up as extensions 2501, 2518, and 2536.
When calling out to another phone, they always
Maybe you can write your own patch that will allow this based on the
useragent somehow mapping it to 2nd peer based on the useragent? But
this feature is not there now.
What will happen when host=dynamic is the last registration will be
the one used, so if you have two SIP devices trying to
appearance (i need to take it off --
SLA currently is very poor, caller id is broken, for one). When I
pickup my Polycom it dials the shared line. When I dial on-hook it
calls without using the shared line.
On Wed, Oct 8, 2008 at 3:58 AM, Vieri [EMAIL PROTECTED] wrote:
--- On Tue, 10/7/08, Andrew
]'
Method: ACK
On Mon, Oct 6, 2008 at 8:26 PM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Tue, Oct 7, 2008 at 2:22 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
The odd thing is on this particular phone it only happens when you
call voicemail.
It is certainly a bug in Asterisk, not the UA. Asterisk
Make sure they are not using double NAT. Many ISPs these days send
their subscribers a modem that in reality is a router.
Also if you can post the PAP2 configuration. I hope you are using
provisioning.. too bad Linksys makes it possible to obtain that
information.
On Mon, Oct 6, 2008 at 12:40
, asterisk svn head.
If it doesn't please send me a bug report and I'm going to fix it.
Best regards
Daniel.
On Mon, Oct 6, 2008 at 7:04 PM, Andrew Joakimsen [EMAIL PROTECTED]
wrote:
That isn't real T.38 support, it's just Packet2Packet bridging that
works correctly. Still need to use a Cisco
I recently did something similar using fax1.com. If you can send an
email you can send a fax that way.
On Tue, Oct 7, 2008 at 9:19 AM, Riccardo Cupardo [EMAIL PROTECTED] wrote:
Hi all,
i wrote a script agi, sking for a code, after that it sends an email now
i need to send a fax... any
The value is not Authenticate ID; From the config file:
# Authenticate ID
P36 = 8000
# Authenticate password
P34 =
If you look at the HTML source of the webconfig the form field you
need to edit will be marked P34.
On Tue, Oct 7, 2008 at 5:30 PM, Ken Zarifes [EMAIL PROTECTED] wrote:
I
as it is, the bad part is I have to make sure the phones
work there and try to troubleshoot from 3000 miles away.
Any work arounds for a problem because of double NAT? A quick and dirty
solution for them to get their phones working right?
Steve Anness
On 10/7/08 2:12 AM, Andrew Joakimsen [EMAIL
On Tue, Oct 7, 2008 at 6:00 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Philipp Kempgen schrieb:
Klaverstyn, David C schrieb:
Mysql for CentOS 5.2 is the mysql client tools.
mysql.i386 : MySQL client programs and shared libraries.
Does anyone have any other suggestions?
Try making sure you use the r option in your dialstring. You should
*NOT* be answering a ringing channel, as Steve suggested, FWIW (if it
doesn't work any other way that is another story)
On Tue, Oct 7, 2008 at 5:04 PM, Nhadie [EMAIL PROTECTED] wrote:
Hi,
Got this weird problem that the caller
I am not sure if it is possible to somehow invoke a function to pick
up the call via dialplan, if it is a combination of that function and
DISA should do what you need.
On Tue, Oct 7, 2008 at 8:37 AM, Vieri [EMAIL PROTECTED] wrote:
--- On Tue, 10/7/08, Anselm Martin Hoffmeister [EMAIL
What do you do to get that message?
On Tue, Oct 7, 2008 at 8:45 AM, Mr surfit [EMAIL PROTECTED] wrote:
Very new to Asterisk, on my console it says there are 47 bad
destinations...What is the best way to track these down and resolve
them
___
--
That isn't real T.38 support, it's just Packet2Packet bridging that
works correctly. Still need to use a Cisco gateway to support sending
the faxes somewhere on the PSTN. But it does work and it is reliable,
I use it every day.
On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
Maybe it works in more recent versions? I don't know. Anyways this is
getting rather off-topic.
On Mon, Oct 6, 2008 at 2:23 PM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Mon, Oct 6, 2008 at 8:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Hopefully it works. The one in CallWeaver doesn't
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can never check voice mail
After about 30 seconds the call drops with these messagess:
[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950
around it, but not being the world's most
proficient C coder, I'm always afraid I'll break something else. ;)
N.
Andrew Joakimsen wrote:
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can
I've used the smaller ones, I think 8pt with 4pt PoE stuck in drop
ceilings and such to power ORiNOCO APs and never had an issue.
As for the larger switches I've used Linksys SRW224P. I have a few
running for a few years without issues. They have GB uplink but the
individual ports are 100M.
On
Yes, you can set moh in sip.conf or zapata.conf. The options are
mohinterpret= mohsuggest=. I think last time I used them (1.2.x)
they were just moh= but it seems mohsuggest=class will do what
you want it to.
On Sat, Oct 4, 2008 at 2:57 PM, carl Lougher [EMAIL PROTECTED] wrote:
This seems
How much further than 300m? It might be very well possible to just
lower the speed to 10M and just use that If you already have some
quality Cat5 cable between both points it's worth a shot. I support
some sites with this arrangement and I've had to find 10M hubs for
replacement hardware (the
Yes. Disable VAD in your Cisco as Asterisk does not (fully) support it.
On Wed, Oct 1, 2008 at 9:21 PM, Gabriel Ortiz Lour
[EMAIL PROTECTED] wrote:
Hi all,
I'm experiencing problems with VAD activated on a cisco router doing the
bridge between an PBX and de asterisk server. The calls are
Has anyone successfully used the IMAP voicemail storage with Microsoft
Exchange 2003? Can someone provide a working example configuration?
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not actually follow the protocol? Why would
it be different?
On Sun, Oct 5, 2008 at 8:38 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Has anyone successfully used the IMAP voicemail storage with Microsoft
Exchange 2003? Can someone provide a working example configuration
I was going to write a blog once about the non-existent T.38 support
in asterisk hence my purchase of the above domain. It expires in 10
days. T.38 support in asterisk still does not exist but I don't have
any time. If someone wants this domain I will offer it for free and
can send push it to your
On Tue, Sep 30, 2008 at 9:23 AM, Lyle Giese [EMAIL PROTECTED] wrote:
1) a two line phone can register with two different * servers or sip
carriers.
Many phones/ATA with multiple lines only allow 1 server and multiple
registrations!
On Tue, Sep 30, 2008 at 6:29 PM, Lyle Giese [EMAIL PROTECTED]
Could someone please tell me where to download Polycom 3.1.0RevB?
Polycom.com is not possible. Thanks.
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Register Now:
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can never check voice mail
After about 30 seconds the call drops with these messagess:
[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950
On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
It is completely illegal in any country that recognizes patents.
You mean countries that recognize software patents, right?
Please do NOT discuss ways to use unlicensed codecs on this list or any other
forum
I have many of the Intel PCI modems in the field working for some
time, but I am trying to find a source for more of them. IMO places
like x100p.com are a rip off -- $40 for a PCI modem? I recall getting
the AMI modems a few years ago for $10. So does anyone know
where I can find the PCI
On Sat, Sep 27, 2008 at 6:52 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
Hi Guys
On the website, we already accept credit card by sending users to paypal
website where we have an account.
PayPal does have a service that is more like a traditional merchant
service. I don't know if they have a
On Thu, Sep 25, 2008 at 7:34 AM, Rizwan Hisham [EMAIL PROTECTED] wrote:
The fax is originated from a fax machine connected to an ata which supports
t38.
That would be great if Asterisk had true T.38 support. It can pass the
T.38 packets it receives to another SIP endpoint (it will do this even
On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
ATAs work OK I guess, just make sure to use a loss less codec such as ULAW.
Since the OP stated he is using E1 lines then he should probably be
using alaw instead.
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When I get a voice message from an unknown caller it will say Message
from telephone number and just not say any number. I was wondering if
I can manually set the caller ID in this case to be something that the
Voicemail app will recognize so it will read out Message from an
unknown caller
I have two sites running Asterisk PBX. Normally the inbound calls go
through a 3rd (colocated) server and are routed via IAX to the site
(the site registers with the main server)
I created a macro that tries to ring one location and then another.
Each site explicitly Answer() the call even though
Does anyone know who maintains the asterisk packages in the openSUSE
buildservice? They are not updating Zaptel with their kernel updates
and I want to get that matter corrected.
I submitted to them a bug report but they seem to not care...
https://bugzilla.novell.com/show_bug.cgi?id=407408 ...
Does anyone know of a spam filter that will work with Asterisk?
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, ISDN, SIP or IAX (etc)
If the telemarketers followed the laws this would not even be a concern.
On Mon, Jun 30, 2008 at 12:16 PM, Brian J. Murrell
[EMAIL PROTECTED] wrote:
On Mon, 2008-06-30 at 12:03 -0400, Andrew Joakimsen wrote:
Does anyone know of a spam filter that will work with Asterisk
Right now the issue I see is you are using overlapping extensions
so maybe that's not working as expected?
you have in context sipura line exten 201, exten 201 included from
context spa and also exten 2xx included from context spa.
What you want to do with sending calls elsewhere if they are
On Fri, Jun 6, 2008 at 9:03 PM, OCG Technical Support [EMAIL PROTECTED] wrote:
Has anyone create the necessary config/kbd file to allow the DiNovo mini to
work well with myth? (Mapped all of the multimedia buttons etc)
Is that in extensions.conf or chan_dinovo.conf?
On Feb 12, 2008 10:40 AM, Ian [EMAIL PROTECTED] wrote:
Hi all,
its been quite a busy few day with pc's packing up etc, I recompile my
whole asterisk today using zaptel 1.4.7.1 and now the problem is
miraculously fixed, I will be sending this report to Digium bugs as well.
Just a quick
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