Philipp von Klitzing wrote:
>> I would like to know if any one have experience with live audio
>> streaming like 1. Streaming from an online resource
>
> Look at app_ices and icecast.
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Ices
If that doesn't work for some reason (In my case
Jonathan Addleman wrote:
> However, I can't find any way to interact with an existing confbridge
> conference. Surely there's some equivalent to meetme's 'meetme list'
> command? Anything else I can use through the cli or manager API? I just
> need to
I've just started switching my project to use confbridge instead of
meetme and app_conference (because of audio glitches that kept appearing
in those applications).
However, I can't find any way to interact with an existing confbridge
conference. Surely there's some equivalent to meetme's 'meet
Jeff Brower wrote:
> Jonathan-
>
>>> How did you measure the gaps? Using signal or speech analysis
>>> software to display the recording? If you measure number of
>>> samples between the gaps, does it correspond to multiples of RTP
>>> packet p
Jeff Brower wrote:
> How did you measure the gaps? Using signal or speech analysis
> software to display the recording? If you measure number of samples
> between the gaps, does it correspond to multiples of RTP packet
> payload length (for example, for 8 kHz G711 multiples of 80 samples
> betwee
marco.mo...@gmail.com wrote:
> It looks to me that u are having clock synchronism problems due to
> the fact you are using Virtual Machine so u don't have an ISDN card
> generating clock. Are u using what was called ztdummie as clock
> source? Can't precise the name of it in chan_dahdi but u have i
I'm having a problem with conferences both meetme and app_conference,
though I've done most of the testing with meetme.
Essentially, I get little gaps in the audio - usually fewer than a dozen
or so samples, though it does vary. They seem to occur at random, but I
usually get one ever few secon
Ian Murray wrote:
>
>>
>> Forgive the possibly stupid question, but do these problems you describe
>> apply equally to the dom0 as to any domU's in a xen system? I used to
>> think not, but now I'm starting to realize that I'm probably mistaken...
>
> Dom0 is still a virtual machine, so I would
David Backeberg wrote:
> Timers are built on the premise that they have access to either a real
> timing device, or unobstructed access to a processor which clicks
> through a proc cycle at a pre-determined rate. Once you break those
> rules, don't be surprised when the timers stop working, and 'ba
me questions / things to try:
What exact version of 1.6.1 are you using?
Are you using the 'default' voicemail context? If not, do you have
'searchcontexts' enabled in voicemail.conf?
Does it work if you add a dummy mailbox to voicemail.conf in the
'default' con
during the record. Is there something I need to enable besides
setting "transmit_silence_during_record=yes" to enable some RTP traffic
outbound during the record?
Jonathan
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I'm still unable to do much with my new 1.6 installation. I just tried
reinstalling, and using the standard debian configuration files, with
just the necessary modifications, in case I had some legacy stuff in
there from earlier versions that was interfering. I'm testing in a xen
domU with debian's
In an attempt to fix problems with EAGI delays in 1.4 (see my other
message for more on that), I've tried upgrading to 1.6, in case it's a
bug that's fixed in the newer version.
Unfortunately, I'm having all kinds of trouble with this new install. My
system relies on conferences, and whenever I
Hello,
I made a post to the forums
(http://forums.digium.com/viewtopic.php?f=1&t=72901&sid=3d5c2717ca5ab7ad676957ae436d4b51)
but haven't received any replies, so thought I'd try here.
On my debian machine running asterisk 1:1.4.21.2~dfsg-3, I've been
noticing that there's a problem with confe
tech department at all for that matter)...
I haven't run iperf through them, so I don't have any performance
statistics. No one has complained except for our fiscal department,
the phones do come at a premium
Hi everybody,
I would like to use realtime authentification with my LDAP.
My Asterisk is v. 1.6.1.12. I'm using OpenLDAP
The command realtime ldap status is OK.
I have configure these files :
/etc/asterisk/extconfig
/etc/asterisk/res_ldap.conf
/etc/asterisk/extensions.ael
I do nothing and I hav
You need to set: host=dynamic Otherwise only .112 is allowed.
-Jonathan
On Tue, Jan 19, 2010 at 1:17 PM, Bruce Ferrell wrote:
> I'm using realtime sip peers and I need to enable a range of IP
> addresses for a peer.
>
> I have:
>
> deny = 0.0.0.0/0.0.0.0
>
to do with
echo on the lines. I would not recommend using them, but YMMV. I do
know that they make good paperweights or dust collectors =)
-Jonathan
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as
fore the
secretaries found my cubical... Fax didn't work the majority of the
time, but if you don't need to send multiple pages all day long it
*might* work for you. An AudioCodes MP-114 costs more, but saves in
frustration and lost of time for those who use the
The best document is the two page quick start guide that came in the
box. You want 5.6, and 5.8 should be out soon if you are an early
adopter.
-Jonathan
Sent from a mobile device.
On Dec 27, 2009, at 9:02 AM, Joseph wrote:
> What what everybody says, it is a good hardware
the Channel to phone number mapping:
Configuration -> Protocol Config -> Endpoint Number -> EndPoint Phone Number
Configure the Hunt group settings
Configuration -> Protocol Config -> Hunt/IP Group -> Hunt group settings
Hope that helps. These are great devices, once you figure
I can't speak specifically to
Caller-ID on FXO ports, as I mainly use them for FXS and local 911
gateways.
-Jonathan
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had to replace them. I put a AudioCodes
MP-124 in and have had no complaints since.
-Jonathan
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/1)
>> -- Executing [...@local:2] VoiceMail("SIP/*15-0849a370", "*11") in new
>> stack
That last line should look like (from my 1.6.1.1 system):
-- Executing [...@local:2] VoiceMail("SIP/*15-0849a370", "*11","u")
in new stack
Did
n the sample
extensions.conf file under the stdexten subroutine.
There are lots of reasons to let the admin decide which greeting to
play. For example, my canned 'receptionist' context plays the busy
greeting as the after-hours greeting, otherwise playing the
unavailable greeting.
-
g the source?
-Jonathan
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a very useful vendor-agnostic protocol.
-Jonathan
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but I'll keep looking
when I get a chance.
-Jonathan
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ad the user info on it,
then kick off the script to change the background image. Might be a
little tricky, but no reboot required!
-Jonathan
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dules for 1.6.1.4 as the latest
>> available.
>
> Yes, there has been progress. The new modules are undergoing testing in
> Digium's Product Quality department and (should they not have any
> regressions) will be released next week.
Any chance
d see if you get any errors around SIP?
-jonathan
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s.
> It might be a good replacement for Linksys.
Not likely. Cisco works great with CallManager, but seems to be
somewhat broken with anything else... wonder why? If you want
something that is dependable and easy to configure I have had great
success with the AudioCodes
pt for the
794x/796x models. You can download the correct SIP software from
Cisco, but are required to have the correct licensing and SmartNet
coverage.
-Jonathan
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digits from the provider. T is useful for outbound calls
with a trunk number such as 9T because you never know what number
those crazy users will try to call.
-Jonathan
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As
ward-dial
> port 0:D
> !
> sip-ua
> retry invite 3
> retry response 3
> retry bye 3
> retry cancel 3
> timers trying 1000
! Don't need this, since you specified it on the dial-peer
! sip-server ipv4:IP_OF_ASTERISK
> !
Good luck!
-Jonathan
_
ou just lock down what
numbers can be called on your PBX.
-Jonathan
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asterisk-users mailing
udioCodes
is about the same cost as a new Cisco solution, but the Adtran would
probably be a lot less. I haven't had a chance to play with Adtran
and Asterisk, but you can register at their website and play with all
of the CLI / GUIs for al
-32
If you have a specific router in mind, I can be more specific.
-Jonathan
On Wed, Oct 14, 2009 at 11:00 AM, Julian Lyndon-Smith wrote:
> I was thinking of putting a cisco router on the E1 line for my test
> system, so I can have multiple test servers accessing the ISDN, rather
&g
just takes you a few more
minutes up front to read the manual.
I haven't tried any Adtran devices but have thought about purchasing
one to test with if I ever get the time.
-Jonathan
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all to my
voicemail extension (3131). Does the voicemail app record video as I've
seen stated in several places? Am I missing something big here?
Thanks in advance
Jonathan
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thing that we do differently is disable VAD on
the phones.
Never used 1.4, only the 1.6 branch. Glad to see that you got it working.
-Jonathan
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some analog handsets.
-Jonathan
On Mon, Oct 5, 2009 at 2:14 AM, Olivier wrote:
> Hi,
>
> In this
> http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/204725/focus=221375
> dating from 2008, experiences with Grandstream GXW4024 were asked.
> Has anyone something up-to-d
; allow the handsets to reinvite each other. Here's the sip.conf
> snippet if it helps:
>
That all looks fine to me. What do your SIPDefault.cnf and
SIP.cnf files look like?
-Jonathan
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asterisk spool, lib, and config files on the DRBD then
> you shouldn't lose voicemail or any configuration.
If someone is in the middle of recording a message, and the server
fails, you will probably lose that message. That's all I was getting
at.
-Jonathan
___
S Repo is somewhat broken for
Cluster... but there is a work around on the bug tracker for CentOS.
Hopefully that will be resolved soon.
Let me know off list if you need any help!
-Jonathan
On Fri, Oct 2, 2009 at 10:58 AM, James Hankins
wrote:
> I'm looking into doing an HA setup for a
hem to the packages section, or download the
tar.gz file in the post script, and auto-compile. However,
auto-compile might have different results on different systems (hence
why I use custom RPMs)
-Jonathan
On Thu, Sep 17, 2009 at 4:27 PM, Neeraj Chand wrote:
>
> Hi guys,
>
> Anyone don
hand, or could reproduce it. Thanks
-Jonathan
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oogle so far, and I think I'm just
not asking the right question.
Thanks for any help or pointers.
-jonathan
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I'm attempting to link calls together in my CDR and would like to try to do it
via the userfield. Is there any way to copy the userfield between calls when
doing an attended transfer? I can't seem to find anything about it searching
Google.
-Jon
___
I have a SIP trunk between CCM 6.1.2 and Asterisk 1.6.1.1 working
without any issues. What does your peer section of the sip.conf look
like? When do you get the error (call direction)?
-Jonathan
On Fri, Sep 4, 2009 at 12:00 PM, Jerry Geis wrote:
> Hi all
>
> I have asterisk 1.4.12
nd often post from here.
-jonathan
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You could put something into the Asterisk Database with DBput/DBget.
I don't have an example off hand, but create a "stickypark" family and
store which channels go back into which parking slot. Or something to
that effect, and it would exist until you remove it from the database.
is ms1.datagrama.net what you really want though? It looks like
you're using "mydomain.com" as the domain in your asterisk
configuration. Do you really intend to use mydomain.com ?
-jonathan
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ed to get things to work correctly could go away entirely.
Good point. It would be much more effective to improve the realtime
integration. Is there a list somewhere of all of these cases? With a
little direction I would be willing to work on this.
-Jonathan
__
tabase base on a record
modification date. This might make it appear more like a static sip
configuration, just using the database for storage. It would also
seem to make it more 'realtime' if things were bidirectional.
I haven't looked that much into the realtime code, but this c
-2MFT-T1 and be easier
to configure.
I will say that when I need to replace my other 2800s I will probably go
with a AudioCodes M1000 (PRI and Analog capabilities), but I don't have any
hands on experience with that device at this time.
-Jonathan
On Thu, Aug 13, 2009 at 4:26 AM, Steve Totar
On Wed, Aug 12, 2009 at 12:39 PM, Olivier wrote:
>
>
> 2009/8/12 Jonathan Thurman
>
>> I am also using them quite extensively, but with English menus. I know
>> that the Locale files from Cisco do not come with the firmware, but usually
>> as an update for Ca
Cisco. You may be able to purchase SmartNet on the phones and get it that
way, or at least they would listen when you called them...
-Jonathan
On Wed, Aug 12, 2009 at 5:23 AM, David Gibbons wrote:
> I am using the phones quite successfully, though I have not tried
> non-English
n handle easily using the soft buttons. Then you could
reuse the extra buttons as speed dials, other specific extensions (i.e. the
boss' DID) etc.
-Jonathan
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== control RTP == voice
in a very generic sense?)
I plan to take a packet trace in the morning on the asterisk server and
see what is going on at that level. Hints as to what I should be looking
for?
-jonathan
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originate on the asterisk server.
I've tried adjusting canreinvite= in sip.conf in hopes in might have
some effect, but so far nothing.
Suggestions on where else to look, or what the problem might be?
Which configs would be useful in troubleshooting?
Thanks.
-jon
ded.
Those lines are like this..
8100 => 1234,Jonathan
8500 => 1234,Support
-jonathan
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cator. (as to my last email, sip set debug does
what I think I need).
When I add two mailboxes, I never see the message in debug about waiting
messages.
So, it seems I was mistaken earlier, and I'm not get MWI events when two
mailboxes are configured as I thought I was.
On Wed, Aug 5, 2009 at 2:55 PM, Doug Lytle wrote:
> Jonathan Moore wrote:
>> Just changed it. Reloaded asterisk and restarted the phone. Same behavior
>> as before. Well, only a single mailbox shows up anyways.
>>
>
> Add the @context on each of the mailboxes:
>
On Wed, Aug 5, 2009 at 1:47 PM, Jared Smith wrote:
> Have you tried 8100&8150 (using an ampersand instead of a comma)?
Just changed it. Reloaded asterisk and restarted the phone. Same behavior
as before. Well, only a single mailbox shows up anyways.
-j
telnet into the phone and see what is going
on. Commands like "show config" "show register" etch are very useful for
this kind of troubleshooting. If the phone was attached to a CallManager
using SIP before, then there could be some bad configuration still stuck in
the phone. If
HN]
type=friend
username=john_smith
secret=**
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=internal
callerid="john smith" <103>
deny=0.0.0.0/0.0.0.0
permit=10.10.0.12/255.255.255.255
-jonathan
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something like
> this
>
> Admin - Music1
> Contrallors - Music 2
> Technical Support - Music 3
Seems like a perfect use of SetMusicOnHold..
<http://www.voip-info.org/wiki/view/Asterisk+cmd+SetMusicOnHold>
-jonathan
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give you a chance to free up some space before Asterisk is affected.
Couldn't you get the same effect using quotas? Also, using separate
partitions for various parts of the filesystem is a nice addition. Having
your /var/log somewhere besides the same partition as / helps keep
runaway logs at ba
ean
>
>
I believe he means that:
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz
is not the same as
svn checkout http://svn.digium.com/svn/asterisk/branches/1.6.0
Which is true as there are lots of things that have
u know, so how to let the gnugk respond and not the asterisk
> h323 channel?
>
Right. If they both run on the same ip/port then the one started first
would win, and listen for connections (the second app should fail to bind
and complain). You could change the port, or the IP that the one
uld
start by looking at the patch in this issue:
https://issues.asterisk.org/view.php?id=15162
Please note again that that patch was against 1.6.1.0.
-Jonathan
On Tue, Jul 14, 2009 at 11:09 AM, Barry L. Kline wrote:
> John A. Sullivan III wrote:
> > Hello, all. I'm having a nasty
On Tue, Jul 14, 2009 at 12:44 PM, VIP Carrier wrote:
> what ever he have posted there I have added it, just changed DID
To help clear things up... what file did you add this to?
-jonathan
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I have one Cisco 7940
configured for Skinny. It seems to work just fine, no seg faults. Have you
tried the latest SVN for 1.6.0?
You should take a look at this issue if you haven't already:
https://issues.asterisk.org/view.php?id=13777
-Jonathan
staying with skinny on these phones? I have
quite a few 7940/7960 converted to SIP that work great.
Next week I will try and duplicate this behavior on my test system with
skinny, but you should get a bug report filed with the core and important
configurations.
-Jonathan
__
ng to avoid adding additional servers at this small sites. Some
sites are nothing more than a portable with Metro Ethernet connection and a
fan-less router and switch.
>
> If they are not, having IT and Telephony to share the same backup WAN is
> advisable.
>
phones to register to in case of WAN failure with 1 or
2 POTS lines attached (also used for 911 calls from that site). Thanks for
any suggestions!
-Jonathan
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.
> But of course, this might be enough for some (operators, ...) but not for
> all (group secretary ...).
No disagreement. It all comes down to how much you're willing to pay for
the convenience. If you (or the users) don't want to switch screens, you ge
>
>
> This has been fixed in the 1.6.1 SVN, and you will have to back port a
> patch until these changes are rolled into another release. I was
> disappointed that more bug fixes were not included in 1.6.1.1.
>
> -Jonathan
>
>
>
> Asterisk 1.6.1.1 was released f
n.
>
This has been fixed in the 1.6.1 SVN, and you will have to back port a patch
until these changes are rolled into another release. I was disappointed
that more bug fixes were not included in 1.6.1.1.
-Jonathan
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bove and the screen, and 6 below. The top
set can have up to 10 configurations, when you add more than 6, the bottom
right button changes to "Next.." and scrolls the screen over. The bottom can
have up to 20, with the same next button.
tus of the other
phones to show up on the side car.
I found all the information on voip-info.org for how to do it. Parking, hints
the aastra configs and all.
You can also do that with the softkeys on the top and bottom of the screen.
-jonathan
___
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want to assign
the fax a number out of your DID block... If you are not a small
office, there are a lot of reasons to not have a dedicated fax PSTN
line.
-Jonathan
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; as the setting.
You can also clear all of the cached settings by telnetting into the phone,
clearing the config, and resetting it.
-Jonathan
On Thu, Jun 25, 2009 at 8:29 AM, David Gibbons wrote:
> Mike,
>
> 1. Remove the 'line 2' entries completely from the SEPXX.XML fil
D"
For each line that you don't want anymore. So on a 7960 you would have to
do this for lines 2-6. The line will then disappear from the phone.
-Jonathan
On Wed, Jun 24, 2009 at 2:11 PM, Mike wrote:
> Folks,
>
> I have CISCO 7940g phone. I have in the past configured th
/products_tech_note09186a00800941bb.shtml
-Jonathan
On Tue, Jun 23, 2009 at 2:53 AM, Sasa wrote:
> Hi, also with your template I have always the same problem !
> Thanks.
>
> --
>
> Salvatore.
>
>
> - Original Message -
> From: "David Gibbons"
> To: "'As
shouldn't need the tlv file.
-Jonathan
On Fri, Jun 19, 2009 at 8:25 AM, Sasa wrote:
> "David Gibbons" wrote:
> > I've found that different types of TFTP servers return differing errors
> > when a file doesn't exist. You don't need the TLV file, but >
contact
me off-list. This is probably one of the best features added to
app_voicemail for 1.6.
-Jonathan
On Thu, Jun 18, 2009 at 11:40 AM, Darrin Henshaw wrote:
> As usual my manager comes up with some obscure reference I didn't find.
> There seems to be a parameter called minpa
tion
> against that because you would almost certainly break something else.
Is that a configuration change or a "let's to edit source code" change?
-jonathan
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I'm not sure if the kernel timing HZ has anything to still do with
things anymore. You might need to recompile your kernel with HZ=1000
-Jon
lf...@leurent.eu wrote:
> Hello all, I have a TC400B Digium card in order to deal with
> transcoding and I have some trouble using it, I have a timer
> s
ant solution!
So if cell phones never require 11 digits...
The company line about NANP and consistancy:
*"We don't care.**We don't have to.**We're the phone company."*-Jonathan
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est version. To put it into Microsoft terms... the minor version is
like a service pack. So CentOS 4.7 is really a base lined version 4,
service pack 7. You get the new features in major releases (like there are
no more "smp" kernels in 5 to deal with)
-Jonathan
On Wed, May 20, 200
that had issues with with having something like
Dial(SIP/remotehost) would fail to connect to remotehost.
-jonathan
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.org and google have told me what I want to doesn't
work, but can't find anything good on what does work. Much appreciate
your guidance.
-jonathan
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names, and then announces it in the
conference on join and leave...
-jonathan
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;, I'd love so pointers
on where to find info on creating audio for this type of project. So
far, google has left me without much. Any help would be appreciated.
Short version: I need to know where to look for some help on recoding
audio for use as asterisk prompts using audacity (or an
ven so, a quick google goes a long way.
-jonathan
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ce line, and with the later
> software echo cancel works OK. Not nearly as bad as some have made it
> out to be, though for US/Canada lines. Not suitable for UK and others
Ah, yes. Thanks for correcting me on that, I was getting some things
mixed up
On Wed, May 6, 2009 at 8:47 PM, ContactTel Business
wrote:
> I'd say in life you get what you pay for.. and sometime you even pay for
> stuff that should be free..
I have to agree.
I have a few of these cards I started out with. They were great for
the "wow, I finally got asterisk to do somethi
how it was. All of this is assuming that you
have a standard CallManager environment of course.
-Jonathan
On Mon, May 4, 2009 at 3:14 PM, David Shauger wrote:
> David,
> Will it happen automatically when you reconnect it to Cisco Call Manager or
> does it require additional steps?
>
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