The desktop versions of snom support Openvpn, i am not sure about M3 (dect).
Take a tour to their site.
2009/2/12 Frank Bulk - iName.com frnk...@iname.com
Not in the form factor that you would expect.
Can I ask why? Most modern VoFi phones support WPA2.
Frank
-Original Message-
I also experience that problem. Is it a bug?
On Wed, Feb 4, 2009 at 5:53 AM, Mark Michelson mmichel...@digium.com wrote:
Remco Barendse wrote:
1.4.23.1 is quite badly broken and there are no significant new
features
There are no new features at all, actually. What problems are you
Ken I was thinking the same thing tonight. I have a Palm Treo 750 (ATT) that is not used anymore. There is one card to add WiFi to it. After that I need to find a SIP client for Windows mobile 5. Any ideas?These phones are cheap and the WiFi card is about $35-40 + shipping.
-Original
This version will hang up the given extension even if it has multiple
channels open:
asterisk -rx show channels | perl -lane print \asterisk -rx \'soft
hangup @F[0]\'\ if m.SIP/201. | bash
perl is always your friend when needing some programming mischief :)
l.
2009/2/12 Danny Nicholas
Benny Amorsen wrote:
Julian Lyndon-Smith aster...@dotr.com writes:
exten = foo,n,Dial(SIP/1234Zap/G1c/55443322)
and SIP/5432 calls this extension,
is it possible to show different callerid numbers to each of the target
numbers ?
No, but you can do
Vikas topg...@gmail.com writes:
The ISP said that they ran a fiber optic wire to a media box at our
office and from there there is a RJ45 to the switch. They bring no new
equipment to our premises each time we provison a new port. Hence this
upload speed limitation is not due to the copper
Tzafrir Cohen wrote:
snip /
But when the wcfxo module is loaded, it is not loading the oslec module.
There is an oslec.ko in /lib/modules/my-running-kernel-version/misc/oslec/
According to launchpad, oslec should be the default ec now for zaptel.
Anyone got any ideas please?
Grygoriy Dobrovolskyy megaho...@gmail.com writes:
The desktop versions of snom support Openvpn, i am not sure about M3
(dect). Take a tour to their site.
Top posting is annoying. Gmail is broken; maybe I should just killfile
@gmail.com.
Anyway, the M3 won't do openvpn, and it's a fairly
This discussion is not making any sense to me.
Just what type of access product is this?
If you have fiber to the premise and are handed Ethernet from there to a
Cisco switch, it is some sort of Metro Ethernet or NMLI (Native Mode LAN
Interconnection) type product. It could also be framed over
Hey Everyone,
I would like to start testing/playing with PRI channels but I don't have access
to a PRI line. Is it possible to do the equivilent of a crossover between two
PRI Cards (say Digium's TE120P)?
What I was thinking is that I could set one asterisk box up with a PRI card set
as the
Default FreePBX context,
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
include = ext-did-post-custom
include = from-did-direct; MODIFICATOIN (PL) for findmefollow if
enabled, should be
On Fri, 13 Feb 2009 09:18:49 + (GMT), Lee Wilson leef...@yahoo.co.uk
wrote:
Hey Everyone,
I would like to start testing/playing with PRI channels but I don't have
access to a PRI line. Is it possible to do the equivilent of a crossover
between two PRI Cards (say Digium's TE120P)?
Oh--you mentioned in an earlier post that the Cisco switch was installed by
the ISP, so presumably that is something they consider their CPE as well.
You can't rate-limit IP bandwidth on Layer 2 switches, and a Catalyst 2950
does not have a Layer 3 feature set; that only comes with MSFCs on
Hello I recently get a Cisco 7940G IP Phone and I try to make several
things with it and I en counted many difficulties:
1.) I tried to unlock the phone and to set manually IP Address,
Netmask, Gateway etc. I don't get any luck.
2.) I tried to upgrade firmware like they said with tftp server... I
I would like to start testing/playing with PRI
channels but I don't have
access to a PRI line. Is it possible to do the
equivilent of a crossover
between two PRI Cards (say Digium's TE120P)?
What I was thinking is that I could set one asterisk
box up with a PRI
card set as the TE
This phone is currently running the SCCP (Skinny) image. Before you will
get anywhere you need to load the SIP firmware image onto it. The SEP*
configuration files are for SCCP.
After doing that, the phone will start requesting the correct files. You
may need to upgrade through various SIP
On Fri, 13 Feb 2009 09:48:11 + (GMT), Lee Wilson leef...@yahoo.co.uk
wrote:
Alex, thanks for the quick response.
So I can assume from your response this should work. That was easy :-)
I just want to clarify before I got and buy anything the cards are not so
cheap.
Yep, it should
you can get the originate resonse with a function dump_event and the
$asm-add_event_handler;off corse you can do it with a script php.
2009/2/13 Aloysius Thevarajah Lloyd (SunTel Technologies)
lloyd.aloys...@sunteltech.ca
Dear All,
I am originating the call directly to the SIP Provider using
I understand, but i cannot load the new firmware... is any well know method?
On Fri, Feb 13, 2009 at 11:48 AM, Alex Balashov
abalas...@evaristesys.com wrote:
This phone is currently running the SCCP (Skinny) image. Before you will
get anywhere you need to load the SIP firmware image onto it.
Have a look at:
http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080094584.shtml#topic2
On Fri, 13 Feb 2009 12:06:48 +0200, Catalin S. jonsonpla...@gmail.com
wrote:
I understand, but i cannot load the new firmware... is any well know
method?
On Fri, Feb 13,
I'm trying to do the same and have read the mentioned sites. The one item I can't seem to get past is a working TFTP server. What is the easiest method to get one running and what packages in Linux or Windows work best?Thanks for putting up with a Linux newbie. Ronny
--
Hey Gordon,
I was probably going with the Openvox anyway because they were cheaper.
As this is just a hobby, no live environment I wanted to move away from just
playing with SIP/IAX and get proper channels setup.
I was inspired by any earlier test I did at my last company where I was able to
On 2/13/09, Alex Balashov abalas...@evaristesys.com wrote:
On Fri, 13 Feb 2009 09:48:11 + (GMT), Lee Wilson leef...@yahoo.co.uk
wrote:
Alex, thanks for the quick response.
So I can assume from your response this should work. That was easy :-)
I just want to clarify
Matt Florell wrote:
just make sure one side is pri_net and the other is pri_cpe and you
should be good to go.
Ah, yes. Definitely don't forget that one. That trips a lot of folks up.
One side must be in ISDN user emulation mode and the other network.
--
Alex Balashov
Evariste Systems
Web
I thought that, with PRI, it was possible to play using a single E1/T1 port
and a T1/E1 loopback adapter (a RJ45 plug with properly assigned wires which
makes any outgoing call an incoming one on another PRI channel) ?
For lab testing, this kind of trick seems very useful.
I've never tried it yet
Olivier wrote:
I thought that, with PRI, it was possible to play using a single E1/T1
port and a T1/E1 loopback adapter (a RJ45 plug with properly assigned
wires which makes any outgoing call an incoming one on another PRI
channel) ?
For lab testing, this kind of trick seems very
The cable needed for this is a different cable than an ethernet cross over.
I have actually done this same thing today with a Samsung 100 system and
Asterisk 1.4.20.1 and Zaptel 1.4.11 and things work great.
A question of my own:
I know I can emulate the network side of a pri connection, but can
A 2950 can be configured to limit the speed per port...
I guess the ISP here is operating this way because they are out of the way
and have limited bandwidth themselves, so, they are trying to split up the
bandwidth provided into smaller, more manageable chunks to avoid overloading
things at
The hotdesking section of the asterisk book may also be of interest...
d
2009/2/13 David Ruggles da...@safedatausa.com
Some googling lead me to this:
http://hans.fugal.net/blog/tag/astdb
Which looks like it has an answer.
Thanks all!
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
On 2/13/09, Tom Moore tommym2...@gmail.com wrote:
The cable needed for this is a different cable than an ethernet cross over.
I have actually done this same thing today with a Samsung 100 system and
Asterisk 1.4.20.1 and Zaptel 1.4.11 and things work great.
I would again just recommend
The ISP tells me that it is a Metro Ethernet product.
Here is a picture of the box made by Mc Mans tel where the ISP
inputs a fat black wire. I have never heard of Mc Mans tel and
googling it comes up with nothing.
http://www.grmtech.com/blog/wp-content/uploads/2009/02/img_0035_1-300x225.jpg
2009/2/13 Alex Balashov abalas...@evaristesys.com
Olivier wrote:
I thought that, with PRI, it was possible to play using a single E1/T1
port and a T1/E1 loopback adapter (a RJ45 plug with properly assigned
wires which makes any outgoing call an incoming one on another PRI
channel) ?
hey finally i did it. I upgraded the firmware to the latest sip
firmware and now i have the another problem. The requested files are
the following:
---///---
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251
Feb 13 14:33:28 linux-9pg5 atftpd[18825]:
Windows - http://kin.klever.net/pumpkin/binaries
Works great.
k4...@bellsouth.net wrote:
I'm trying to do the same and have read the mentioned sites. The one
item I can't seem to get past is a working TFTP server. What is the
easiest method to get one running and what packages in Linux
Matt Florell wrote:
If you want to be able to set Master timing you will have to use
Sangoma cards, they allow you force the timing in the wanrouter
configuration. I have done extensive testing with crossover T1/E1/PRIs
I don't believe this is true; we use Digium cards connected back-to-back
2009/2/12 Ken D'Ambrosio k...@jots.org
Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN
abilities? Failing that, a WiFi phone that runs Linux? I already know
one phone that does meet my requirements -- the iPhone. The new software
comes with a Cisco VPN client, and
2009/2/13 Kevin P. Fleming kpflem...@digium.com
Matt Florell wrote:
If you want to be able to set Master timing you will have to use
Sangoma cards, they allow you force the timing in the wanrouter
configuration. I have done extensive testing with crossover T1/E1/PRIs
I don't believe
Olivier wrote:
What if using a patch cord from port to another ?
Would you still value to zero each span timing priority or would you set
one end to 0 and the other to 1 ?
For connecting spans on a single card to each other, all the spans
should be set to '0'; there is no need to use a
On Thu, 12 Feb 2009, asterisk_h...@iwishi.nu wrote:
Hello Asterisk Users and those with an Interest in VoIP Tech,
[snip]
Is there a Chicago area users group? If not is there any interest in
creating one?
j
___
-- Bandwidth and Colocation
Lee Wilson wrote:
Hey Everyone,
I would like to start testing/playing with PRI channels but I don't have
access to a PRI line. Is it possible to do the equivilent of a crossover
between two PRI Cards (say Digium's TE120P)?
What I was thinking is that I could set one asterisk box up
Thanks Dave, I hadn't read up on Dynamic Spans before and will certainly take a
look. If anyone else is interested I did a quick google search and came up
with the following links:
http://www.microalcarria.com/descargas/documentos/Linux/varios/Asterisk/asteriskdocs-docbook/docs-html/x1320.html
On Fri, 13 Feb 2009, Vikas wrote:
In my opinion the only strategy that has a high probability of success is:
Get a Cisco with five ethernet ports. Use one for your connection to
asterisk. Use the other four as your connection to the ISP, and MUX them.
Can you please point me to some
I'm moving over to asterisk-dev. Seems to be a bug.
Greetings,
Gunnar Schaller
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
2009/2/13 Vikas topg...@gmail.com
My questions are:
1. The black wire coming into the Mc Manstel box is that a fibre optic
cable ?
2. What is the Mc Manstel box doing ?
3. What CISCO router do I need to buy to do bandwidth aggregation at my end
?
1) Yes
2) It's stopping you from poking
Please send on! Thanks! What TFTP server did you use?
Catalin S. wrote:
hey finally i did it. I upgraded the firmware to the latest sip
firmware and now i have the another problem. The requested files are
the following:
---///---
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
k4...@bellsouth.net wrote:
I'm trying to do the same and have read the mentioned sites. The one
item I can't seem to get past is a working TFTP server. What is the
easiest method to get one running and what packages in Linux or
Windows work best?
Solar winds TFTP server, free, is
Catalin S. wrote:
hey finally i did it. I upgraded the firmware to the latest sip firmware and
now i have the another problem. The requested files are the following:
---///---
Feb 13 14:33:28 linux-9pg5 atftpd[18825]: Serving
CTLSEP00141CAA4B4C.tlv to 192.168.1.3:51251
Feb 13 14:33:28
I have two more questions:
1. Is it a technical reason that the ISP has restricted the upload to
512 Kbps or is it a Marketing reason that they have restricted the
upload ?
2. Can I boot the cisco switch in run level 1 and modify the rate
limits on each of the ports ?
This vidoe talks about
Thanks! I'll have to load it on my Wife's Windows machine. I have gone
total Linux and the darn things keep going! My wife's Vista WinBlows
laptop has to be booted every few days.
pe...@networkoblivion.com wrote:
Windows - http://kin.klever.net/pumpkin/binaries
Works great.
You guys think YOU'RE overdoing it... your solution works with a single line.
My solution was some convoluted 100 line shell script!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Lenz Emilitri wrote:
I have a feeling we're overdoing it :)
l.
If you are running Linux you already have a TFTP server. You just need to
enable it.
j
On Fri, 13 Feb 2009, Ronny Julian wrote:
Thanks! I'll have to load it on my Wife's Windows machine. I have gone
total Linux and the darn things keep going! My wife's Vista WinBlows
laptop has to be
On 2/13/09, Kevin P. Fleming kpflem...@digium.com wrote:
Matt Florell wrote:
If you want to be able to set Master timing you will have to use
Sangoma cards, they allow you force the timing in the wanrouter
configuration. I have done extensive testing with crossover T1/E1/PRIs
I don't
On Fri, Feb 13, 2009 at 9:58 AM, Vikas topg...@gmail.com wrote:
What would do if you found yourself in such a situation ?
I would switch to a phone company phone line. An E1 or T1.
It sounds like this company is a data provider and not a phone company.
The solar winds windows server is very nice while troubleshooting
though, as you have an on screen log that gives real time reports if
something is in error.
Once all is working, have the Linux TFTP available for the phone when it
needs it.
And sometimes enabling and KEEPING it enabled can be a
Jeff LaCoursiere wrote:
If you are running Linux you already have a TFTP server. You just need to
enable it.
So there is one on the Asterisk box then? How would I enable it there?
Thanks for helping a newbie!
___
-- Bandwidth and Colocation
On Fri, Feb 13, 2009 at 07:49:05AM -0600, Kevin P. Fleming wrote:
Olivier wrote:
What if using a patch cord from port to another ?
Would you still value to zero each span timing priority or would you set
one end to 0 and the other to 1 ?
For connecting spans on a single card to each
On Friday 13 February 2009 07:54:48 Jeff LaCoursiere wrote:
Is there a Chicago area users group? If not is there any interest in
creating one?
there is: http://groups.google.com/group/asterisk-chicago
though it's fairly inactive.
--
Anthony - http://messinet.com -
On Fri, Feb 13, 2009 at 10:23:04AM -0500, John Novack wrote:
The solar winds windows server is very nice while troubleshooting
though, as you have an on screen log that gives real time reports if
something is in error.
Hint:
tail -f /var/log/daemon.log
(on Debians. probably
On Fri, 13 Feb 2009, Ronny Julian wrote:
Jeff LaCoursiere wrote:
If you are running Linux you already have a TFTP server. You just need to
enable it.
So there is one on the Asterisk box then? How would I enable it there?
Thanks for helping a newbie!
Depends on your flavor of Linux and
All;
I wrote a PERL AGI script that prompts a caller to leave a message using
print RECORD FILE $recordfile wav # 6 BEEP s=3\n;
When the caller is done, they need to press the # key. The message is then
delivered.
However, the message is not delivered if the caller simply hangs up when
I'm just a tad north of you up here in Duluth. However, you just *had* to pick
Valentines Day for the meetup didn't you? Ugh. I'd love to make the drive and
come down to discuss Asterisk but unfortunately the *other* love of my life
wouldn't quite agree with that. :-)
Do you have any sort of
I am seeing this problem on 1.6.0.1 when dialing a busy DAHDI channel...
On Fri, Feb 13, 2009 at 8:40 AM, Rilawich Ango maillist...@gmail.comwrote:
I also experience that problem. Is it a bug?
On Wed, Feb 4, 2009 at 5:53 AM, Mark Michelson mmichel...@digium.com
wrote:
Remco Barendse
Quoting Tim Nelson tnel...@rockbochs.com:
Do you have any sort of site/mailing list/etc setup to facilitate
this group? I'd be interested in attending such a meetup in the
future.
http://www.tcaug.net/
1. Is it a technical reason that the ISP has restricted the upload to
512 Kbps or is it a Marketing reason that they have restricted the
upload ?
Marketing most likely. If they have fiber in the building, it would be
a symmetric link. I have never seen a fiber connection that isn't the
Voxilla published a two-part series on using Amazon’s cloud services
to meet your business telephony needs. Part 1 covers Amazon EC2 and
how it is used in a VoIP setting. Part 2, covers all the steps
necessary in getting the open-source Asterisk PBX to work on Amazon’s
cloud.
On Friday 13 February 2009 09:55:49 cbbs...@hotmail.com wrote:
All;
I wrote a PERL AGI script that prompts a caller to leave a message using
print RECORD FILE $recordfile wav # 6 BEEP s=3\n;
When the caller is done, they need to press the # key. The message is then
delivered. However,
Anybody here is able to use Aastra phones with Asterisk 1.6.0.5?
Making calls is not a problem but when you receive a call it always
drops at 1:45 minutes, always!
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
Matt Florell wrote:
Can you tell me where the setting is to force Master timing on Digium
cards per port? I really didn't think Digium cards had the ability to
force Master in this way. I've tried to do it with channelbanks before
and couldn't force it to master, whereas I can get it to work
Tzafrir Cohen wrote:
But this is not the case if both are on two different cards.
In addition to that, does a T1/E1 link work well when nither side
attempts to set timing?
You are correct; if the spans are on different cards, then one of them
would ideally be configured to use the clock
I've been involved with getting better data for running Asterisk on
the Amazon EC2 cloud computing system. Here are some calculations
I've made on costs based on current published prices on Amazon's
system. Feel free to tell me that I'm wrong with these calculations -
but be specific if
I've got SIP load SIP41.8-4-1S running w/o problems in a stable environment.
I'll provide SEPMAC.cnf.xml's if requested off-list.
--Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent:
On Friday 13 February 2009 11:39:07 Carlos Chavez wrote:
Anybody here is able to use Aastra phones with Asterisk 1.6.0.5?
Making calls is not a problem but when you receive a call it always
drops at 1:45 minutes, always!
I use 1.6.0.5 with 3 Aastra 480i CT phones and have no issues
On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
I've been involved with getting better data for running Asterisk on
the Amazon EC2 cloud computing system. Here are some calculations
I've made on costs based on current published prices on Amazon's
system. Feel free to tell
David Gibbons wrote:
I've got SIP load SIP41.8-4-1S running w/o problems in a stable environment.
I'll provide SEPMAC.cnf.xml's if requested off-list.
--Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Hi there,
is gizmo the first user of the Digium Skype solution, or do they use a
different approach/product - any clue?
http://www.gizmo5.com/pc/opensky/
Philipp
___
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On 2/13/09, Kevin P. Fleming kpflem...@digium.com wrote:
Matt Florell wrote:
Can you tell me where the setting is to force Master timing on Digium
cards per port? I really didn't think Digium cards had the ability to
force Master in this way. I've tried to do it with channelbanks before
On Fri, Feb 13, 2009 at 8:19 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de is gizmo the first user of
the Digium Skype solution, or do they use a
different approach/product - any clue?
No clue how it's done, but it cuts off at 5 minutes and the quality
isn't fantastic
On Feb 13, 2009, at 11:05 AM, Tzafrir Cohen wrote:
On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
I've been involved with getting better data for running Asterisk on
the Amazon EC2 cloud computing system. Here are some calculations
I've made on costs based on current published
On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote:
Hi there,
is gizmo the first user of the Digium Skype solution, or do they use a
different approach/product - any clue?
http://www.gizmo5.com/pc/opensky/
Philipp
I know nothing about their solution that I can say with assurance
snip
On a similar subject, I have been able to get a 7961 to switch to a SIP
firmware, has anyone had any luck with this?
/snip
Yes, I have several 7961s and 7971s running SIP, same firmware generation as
the 41s
--Dave
___
-- Bandwidth and
Matt Florell wrote:
My understanding is that this setting lets you ignore the timing
signal coming from the other end of one of the ports, and the card
will take another timing source from a port that you specify and
force it to be used as a timer on that first port.
That is very close to
On Fri, Feb 13, 2009 at 11:35:53AM -0800, John Todd wrote:
On Feb 13, 2009, at 11:05 AM, Tzafrir Cohen wrote:
On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
I've been involved with getting better data for running Asterisk on
the Amazon EC2 cloud computing system. Here are
Hi all:
when i make a call from linksys pap2t to an asterisk server a fake ring is
heard some times ,but when sending calls between 2 asterisk servers through sip
no fake ring is heard but real one.
any suggestions please.
_
On 2/13/09, Kevin P. Fleming kpflem...@digium.com wrote:
Matt Florell wrote:
My understanding is that this setting lets you ignore the timing
signal coming from the other end of one of the ports, and the card
will take another timing source from a port that you specify and
force it
I am new to the Asterisk world, but have decided to use the business
edition, but am looking for a cost effective gui interface to manage the
software. I have heard of FreePBX, ScopServ and Trixbox from Fonality. Can
anyone give me a heads-up on any of these products and possibly a
comparison?
Matt Florell wrote:
When I got it working I had set the clock source to the 4th port on
the card for the channelbank which was on the first port. We never
tried any other ports because I was afraid to touch it after I got it
working.
Understandable; as far as I know, the Digium card should
i finally did it... It works excellent. Thank you guys for help.
On Fri, Feb 13, 2009 at 9:44 PM, David Gibbons d...@videon-central.com wrote:
snip
On a similar subject, I have been able to get a 7961 to switch to a SIP
firmware, has anyone had any luck with this?
/snip
Yes, I have several
The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 1.6.0.6, tagged as version 1.6.0.6-rc1. Release candidate
1.6.0.6-rc1 is available for immediate download at http://downloads.digium.com/.
This release of Asterisk fixes several issues related to overall
Could you please send me a copy of your sip.conf file so I can compare
the settings? Aastra 5xi phones in my office always drop the call after
105 seconds and older type phones like the 9133i and 480i crash after
about a minute. It only happens with Aastra phones and only when
receiving
On Fri, 2009-02-13 at 15:24 -0500, Miguel Martinez wrote:
I am new to the Asterisk world, but have decided to use the business
edition, but am looking for a cost effective gui interface to manage
the software.
Does the Asterisk-GUI work with Asterisk Business Edition?
Bob Pierce wrote:
Does the Asterisk-GUI work with Asterisk Business Edition?
Yes, and it is included with ABE and installed as part of the standard
installation process.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
OCG Technical Support schrieb:
We use extensions like plant201 and tunnel12 so it does work in 1.4
As a *pattern* (e.g. _plant2XX, _tunnel.)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose P.
On Fri, 2009-02-13 at 15:26 -0600, Bob Pierce wrote:
Does the Asterisk-GUI work with Asterisk Business Edition?
Yes, it does.
--
Jared Smith
Digium, Inc. | Training Manager
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Benny Amorsen schrieb:
Top posting is annoying. Gmail is broken; maybe I should just killfile
@gmail.com.
Emails sent through Gmail's *web interface* are broken. :-)
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk:
We made a very simple application to insert the cost of a call into the
CDR table that Asterisk uses. We recently upgraded to Asterisk 1.6 and
I noticed that my application stopped working.
The reason is that my application depends on a field called route to
be NULL so that it
No, sorry, we match _XXX to jump to plant123
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: February 13, 2009 4:35 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Strange
I have had to install a TDM800 in a site, as the telco has held off
installing ISDN indefinitely..
It's all fine except for the fact that it takes ages to hang up the line
(6 or more rings), and sometimes doesn't even bother. This is only on
incoming calls - outgoing calls work perfectly.
Is
Folks,
I've read some sources claiming that Asterisk does not need DAHDI for
timing in 1.6.1. Is this true? Searching the web, all I can find is
pages celebrating the fact but no actual documentation on which version
it was introduced in and how one would go about configuring an external
time
Just google 2950 and rate-limit and you'll see that it's possible to do so
with the EI immagebut in 1 Mbps increments.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Friday,
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