when using MixMonitor(), is
there a better way to do recording considering my particular usage?
Should I try old versions of 1.4, and see if the echo is some recent
regression in the MixMonitor() code?
Thanks for any suggestions, (and of course thanks for an incredible
open-source product)
David
On Mon, May 5, 2008 at 5:44 AM, srinivas Antarvedi
[EMAIL PROTECTED] wrote:
but with a DID inward dialing if i want to call a particular command
of MeetMeAdmin()
exten = DID ,1,MeetMe(confnumber,command,user)
This is an unfeasible solution
Then how can i do that as i cannot create
)
exten = s,5,MeetMe(1234|Mp1r) ; notice the argument 'r', meaning
start recording
exten = s,6,HangUp
exten = h,1,MeetMeAdmin(1234|Q) ; notice the argument 'Q', which
stops recordings if you apply my patch
exten = h,2,HangUp
Happy Asterisking!
David Backeberg
toggle_off_recordings.patch
Tzafrir Cohen wrote:
The only downside is that you can simply concatenate two files using
'cat file1 file2 file1file2' with wav as you can with raw formats
(provided that both originals are of the same format), because the
header is not part of the stream.
Correction for the
Just out of curiosity:
I can't remember when I last had to concatenate 2 sound files.
So why does this always come up? IMHO it's one of those things
you hardly ever need.(?)
It's all about how you define need. Obviously anybody can make
multiple script entries to play multiple files. My
On Tue, May 13, 2008 at 10:42 AM, Asterisk [EMAIL PROTECTED] wrote:
Is there any way to modify the volume (either lower the volume of the
clients, or increase the volume of the agents) while doing the join of the
-in and -out files into one recording?
Uh-huh. Read the documentation for
find out some settings for soxmix, do you maybe know where can I
change Asterisk settings for soxmix (parameters)?
Regards, Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David
Backeberg
Sent: Tuesday, May 13, 2008 5:35 PM
To: Asterisk
No, no, no.
Don't try to play them directly as gsm files. Convert them to wav on
the fly, when demanded by the user from the webpage. Have a php, or
perl, or whatever script call sox, and push the wav to the user. sox
runs so fast that you can do the conversion on-demand. You can decide
what to
I'm interested in how many concurrent calls Asterisk can process without
troubles. I mean 1 Asterisk server (software) like either proxy or media
server (any numbers will be appropriate).
Since one standard answer to this question is: it depends on how
you're using it,
The ideal situation is
Has anybody ever tried to roll their own VoIP or Zaptel load
simulator? How did they do it?
SIPP can help with benchmarking SIP calls and you can loop back T1
calls if you have two machines with T1 cards or even one machine with
multiple T1 ports.
SIPp looks like it's exactly the right tool
On Fri, May 16, 2008 at 6:58 AM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Does anyone know where I can a copy of sox for windows with mp3 built in ?
It might be in the cygwin project, but I've never tried. If it's not
in the project, I'm not surprised.
You could download knoppix and encode
Make sure you enable all the USE flags, and then perhaps try
emerge boost
again
I've had times where leaving out a badly named USE flag meant that
critical things didn't end up getting built. A particularly egregious
must enable all USE flags case is if you try
emerge ejabberd
Without all the
There's so such thing as privacy on a phone call, at least not in the
United States since warrantless wiretapping. Then expecting security
on a conference call, which by definition is open to many parties, is
silly.
Depending on your state (in the US), you may need to disclose when and
if you do
callerid_internal=test 710;did=551234
Again, this works fine. The problem is when I forward my calls to another
outside line (using Polyocm phones), and need to know the ${did} value at
that point. It's empty.
The other answer looks pretty good. If that doesn't work, do a sip
debug on
You can enable debugging and make sure that the assignment earlier in
your dialplan is always assigning a value.
That is, if you assign nothing to CDR(accountcode), of course the
value will be nothing. Is there any chance that there's occasionially
something going wrong earlier such that there's
I'm guessing you're using gentoo. Your LDFLAGS ended up being
backslash. Don't do that, and it will probably build. I'm making an
educated guess, based on your 'omit-frame-pointer'. Don't omit your
frame pointer either, if you ever want help debugging. Go into
/etc/make.conf, find the gentoo docs,
MeetMe() provides very useful tones when a caller is added to a MeetMe
room. That is, if you're using the musiconhold option, the agent would
hear music, immediately followed by two tones, and then they would be
bridged to the client.
Perhaps you're running MeetMe() with those join tones
Do a reverse lookup on your attacker.
Then find their ISP.
Then file an abuse complaint.
On Mon, Jun 30, 2008 at 12:15 PM, spectro [EMAIL PROTECTED] wrote:
Hello, yesterday one of the extensions on my asterisk server got
compromised by brute-force attack. The attacker used it to try pull an
:
On Mon, Jun 30, 2008 at 1:31 PM, David Backeberg [EMAIL PROTECTED] wrote:
Do a reverse lookup on your attacker.
Then find their ISP.
Then file an abuse complaint.
already done, also filed a report with FBI cybercrime unit and setup
iptables to block incoming traffic from that IP.
My
On Mon, Jun 30, 2008 at 5:10 PM, Kristian Kielhofner
[EMAIL PROTECTED] wrote:
Does anyone want to write a kernel module? ;)
The thing I was mentioning about hashing addresses is already in the
kernel, check out:
hashlimit on google,
or net/netfilter/xt_hashlimit.c in your favorite 2.6 kernel
On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote:
Maybe someone can help me to track down the problem. What should I
check, monitor test. Any ideas are welcome.
If there are no legal reasons not to, consider recording all calls for
a limited time. It's easier for engineers
On Mon, Jul 7, 2008 at 1:21 PM, Douglas Garstang [EMAIL PROTECTED] wrote:
So, I need to build a complicated IVR with Asterisk, with a lot of back end
hooks. The dial plan itself has a lot of limitations, not the least of which
is that the dial plan is ugly, hard to maintain, and full of gotchas
On Sat, Aug 9, 2008 at 2:53 AM, Venefax [EMAIL PROTECTED] wrote:
I have to rotate the Master.csv every 5 minutes, on a heavy loaded system. I
use cdr-custom. How do I force asterisk to rename the cdr file every 5 mins?
Is there a way?
Why do you have to rotate the file every 5 minutes?
Why not
The other answer is:
Sip handling is improved in 1.4 and then improved again in 1.6. If you
really grow your sip traffic you should try to find time to upgrade to
1.4
On Fri, Aug 8, 2008 at 4:29 PM, Chris Elliott
[EMAIL PROTECTED] wrote:
I'll answer my earlier question regarding System commands
If you're using IAX, upgrade to latest 1.4, as there have been a
number of security releases you should apply. Or at least read up and
make sure the 1.2 version is up-to-date.
On Wed, Aug 6, 2008 at 1:40 PM, Rosli Sukri [EMAIL PROTECTED] wrote:
hi,
wanted to ask if anybody has experienced
You might want to look at ForkCDR() and ResetCDR() to see if they meet
your needs. Or if you really need the accounting that cisco provides,
bounce your calls through that cisco box first.
Honestly, if your hardware is crashing a lot, I'd worry more about the
root cause of that than the fact that
An educated guess is:
reverse the SIP trunk buttons, so the preferred provider is the top
button, and voila, your speed dial going to the first trunk is now
what you want.
On Wed, Aug 13, 2008 at 7:44 PM, Shawn L [EMAIL PROTECTED] wrote:
This one is a little off-topic, it's more about the phone
This is weird. Any help you can offer would be appreciated. We spent 6
hours on phone with Digium support yesterday and could not locate an
issue within asterisk itself.
Have you tried putting a soft phone on the same machine as the
asterisk box? Put a sound card in there, connect a
I don't know exactly how Zen handles emulating clock interrupts. We've
done well using Asterisk within VMWare, although we had no need for
meetme on those particular machines. There's a lengthy white paper
about how VMWare fakes a real clock, which you can find at:
They want an after hours application that checks inbound caller ID
numbers and matches them to a list, say 5 to 10 numbers of special VIP
customers, if there is a match on the list, then forward the call
straight to a cell phone, instead of ringing local extension and then
to voicemail.
The
The supervisor will have a control panel, where he will see how many
of his agents are on call. If they are, he can right-click on the
agent and get the options Call Monitor (where the super just listens
in on the call, or new reps can listenin), Call Train (where the super
and agent can talk
I'm looking for a GUI like ARI by LittleJohn Consulting (which is not being
maintained actively anymore, but FreePBX seems to include it) so users can
login, check cdrs, recordings, call forward, etc.
What's wrong with just using FreePBX directly?
Because that would mean changing the entire vanilla framework with over 200
users on it.
I don't know anything about your vanilla framework, or how complicated
it is. That said, there are lots of places in FreePBX where you can
interject regular vanilla dialplan, you just end up using files
I have a Cisco 3845 with a ISDN PRI port connected to my legacy PBX, this
router is running IOS 12.4(5) T5. I'm trying to integrate Asterisk with this
router through H.323, I tried ooh323 (comes with asterisk-addons) and it
works partially, I can make calls from Cisco to Asterisk, but the
On Fri, Sep 19, 2008 at 2:29 AM, Hiren Mistry
[EMAIL PROTECTED] wrote:
Hear span 1 is connect with pri which is for outgoing and span 2 is connect
with Samsung OpenServ 500 PBX for PSTN Agent.
Pl. any one can give me help. B'coz I have to implicitly work for
Outgoing call from PSTN
On Tue, Sep 16, 2008 at 3:28 PM, Guilherme Loch Waltrick Góes
[EMAIL PROTECTED] wrote:
We tried to setup SIP between Asterisk and the Router, but the SIP stack in
this IOS version is broken and causes the router to reboot.
My biggest problem is, I can't upgrade de IOS version of the router.
I wonder what this means in the long run for the open development of this
platform?
Not a darn thing, unless Cisco screws around and makes an incompatible
version of a jabber server and client that doesn't play according to
the protocol. Microsoft Java, anybody?
We'll see how long this list
My outbound dialing rule was incredibly complex:
exten = _X.,1,Dial(${PASSTHROUGHTRUNK}/${EXTEN})
And everything seemed to be working ducky, until I went to call Germany
and got -- a local cell phone number. Needless to say, this puzzled me
greatly. A quick look at my log, though, showed
One option might be to run in the opposite vmware direction. That is, run
Linux as the native OS and run Windows within a vmware instance. That
gives you the Windows compatibility for your applications, while at the same
time providing the critical hardware timing for your Asterisk instance.
Isn't IMAP IMAP? Does MS 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
Is there some kind of splitter which, on one side can accept two E1
connections from Asterisks and on the other side one E1 link from PBX. This
splitter must also recognize towards which one of two E1 links on Asterisk
side it should send signals to. eg. when primary Asterisk fails this
my dial plan is like callers dial into asterisk(span1) , hear an IVR
option and they are connected to the agents via the legacy pbx (which is in
sync with asterisk on span2)This works perfectly fine until about 200
calls or so...After that time when asterisk tries to dial to the legacy pbx
those cards don't terminate faxes directly; that is, they aren't fax
modems. You can redirect the call to a fake fax with hylafax and
asterisk.
2009/1/12 Gregory Malsack gmals...@gmellc.com:
Hello All,
I have a need to connect an analog device to an asterisk server. The analog
device has 4
Listed below because they are easy to remember and have never failed
me even when a customer's ISP's DNS is down, I get the call Our
Internet is Down but they can ping my servers by IP. Also, can
speed up complaints of a slow network (if DNS lookups are the reason
for the slowness.
4.2.2.1
I had the same problem doing SIP - IAX, 1.4.19.1 as well as the last 1.4.22
In my case I was trying to do FAX and the blips were breaking lots of
the faxes. My solution was to switch to T.38 over SIP and (cross my
fingers) the problems haven't came back so far.
I don't know the source of the
you have asked several questions that have little or nothing to do
with asterisk. Perhaps you should purchase some consulting time from a
linux admin, join your local Linux Users Group, or at least ask your
questions in a newbies forum for the version of linux you have
chosen...
If you insist on
You seem to have two network devices in that system:
intel, probably onboard
dlink, probably a pci or pci-express card
one is working fine, the other is not. This is usually a kernel driver
problem you will need to work out. Check out the CentOS forums to see
whether your dlink card is supported
On Wed, Jan 28, 2009 at 2:48 PM, Jim Dickenson dicken...@cfmc.com wrote:
I have this extension:
exten = 1322,n,Playback(tt-weasels)
Clearly the problem is that weasels have eaten [your] phone system. :-p
But really, you posted your 1322 condition, and then asked about
something happening with
Has anybody ever asked Digium to provide something like the kernel.org RSS feed?
My impression was this was created because kernel.org was tired of how
many people built cron-ified wget scripts against kernel.org | diff
against last get | sendmail
you get the idea
Perhaps downloads.digium.com
http://lists.digium.com/mailman/listinfo/
consider joining asterisk-ha-clustering
or at least looking through their archives
I do all my balancing via SIP and DNS round robin (and a few other
more custom things). Works quite well for me.
On Thu, Jan 29, 2009 at 8:56 PM, David fire
On Mon, Feb 9, 2009 at 4:23 PM, Olivier oza-4...@myamail.com wrote:
Hi,
I would like to improve my understanding of T.38.
I recommend you try out Asterisk 1.6 if you want to play with T.38.
I DID get asterisk-1.4 working with fax, but I was having a lot of
issues with faxes dropping in weird
On Tue, Feb 10, 2009 at 1:52 AM, Olivier oza-4...@myamail.com wrote:
Have you tried to directly connect two T.38 enabled gateways without
involving Asterisk at all (like this) ?
ISDN --- Gateway --- SIP/T.38 --- ATA --- FXO/FXS --- fax machine
No. As long as you can sufficiently debug that
On Thu, Feb 12, 2009 at 4:04 PM, Vikas topg...@gmail.com wrote:
My three questions are:
1. Is there any technical reason behind why the ISP will not sell more
then 512 Kbps of b/w on a single port to us ?
Yes. Somebody programmed their equipment that way and didn't train
anybody else on Cisco
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.
On Tue, Feb 17, 2009 at 5:00 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Meanwhile what's the 2nd best way to use ztdummy - force it to use the RTC
instead?
The best way to use ztdummy is to read about the change to using
DAHDI, and use dahdi_dummy instead.
On Tue, Feb 17, 2009 at 1:51 AM, Rajkumar S rajkum...@gmail.com wrote:
How can I stress test an asterisk IVR? I am looking for some kind of
sip phone which can be programmed to send out digits after specified
time to simulate users pressing menu items. If it can originate large
number of calls
On Mon, Feb 23, 2009 at 12:06 AM, Joseph L. Casale
jcas...@activenetwerx.com wrote:
I am running Asterisk as non root and have set the required permissions for
all
directories including the moh dir specified in musiconhold.conf yet asterisk
still complains it doesn't have access when
On Mon, Feb 23, 2009 at 3:30 AM, fateme fatah faza_...@yahoo.com wrote:
extensions.conf:
[from-pstn]
exten = 9711315,1,Answer()
exten = 9711315,2,Wait(10)
Why on earth are you waiting TEN seconds to actually receive the fax?
Have you tried ripping that out of your dialplan?
exten =
On Mon, Feb 23, 2009 at 9:10 PM, Vikas topg...@gmail.com wrote:
1. When people call in on the 800 number take the local number they
are calling from and then call them back from our unlimited outgoing
account from broadvoice.
I would recommend IVR-ing this as an option, on the premise that
On Mon, Feb 23, 2009 at 9:10 PM, Vikas topg...@gmail.com wrote:
I have been using the inbound 800 services from vitelity. Slowly the
usage has been rising and in the month of Jan the bill was for $650. I
am currently on a 1.9 cents a minute plan. Am I paying too much ?
I don't pay the bill, so
On Tue, Feb 24, 2009 at 2:59 AM, Rilawich Ango maillist...@gmail.com wrote:
Hi all,
Is it possible to install more than 1 asterisk in a single server?
Can somebody help me understand why you would want to do this?
I suppose development versus production, but wouldn't you also want
better
On Fri, Feb 27, 2009 at 2:45 PM, Daniel Hazelbaker
dan...@highdesertchurch.com wrote:
Is there a way to force a channel to continue in the dialplan after
the remote end hangs up?
You use the 'h' side of the dialplan for the extension.
exten = s,1,Answer
exten = s,n,Set(some magic to make
On Tue, Mar 3, 2009 at 7:12 PM, Dean Collins d...@cognation.net wrote:
I’ll pay anyone a $20 bounty for someone to replicate the USA Asterisk
Weather App on Tropo.
All you have to do is violate the ToS on a few services:
wget the weather from yahoo, for instance:
On Tue, Mar 3, 2009 at 10:25 PM, David Backeberg dbackeb...@gmail.com wrote:
exten = 123,s,1 Playback(enterzipcode)
exten = 123,s,n Read(zip||5)
exten = 123,s,n System(wget http://pathtoyahooservice${zip} -o forecast.txt)
exten = 123,s,n System(wget --post-file forecast.txt -o wav.url)
exten
On Wed, Mar 4, 2009 at 9:11 AM, Bill Michaelson b...@cosi.com wrote:
It's conceivable that the combined effort of these two responders required
less than ten minutes of time, yielding a theoretical pay rate of $120/hour.
I wonder how much effort went into the other responses.
I'm reminded of
On Tue, Mar 10, 2009 at 10:27 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
Hello,
I'm having difficulties to make Asterisk (1.6.0.6) interoperate correctly
with a CISCO mediaGW in order to send faxes to the PSTN using T.38.
Is it possible to make Asterisk work like this?
yes, as I've
On Tue, Mar 10, 2009 at 11:18 AM, Joshua Colp jc...@digium.com wrote:
- Santiago Gimeno santiago.gim...@gmail.com wrote:
This was filed as an issue and is being tracked at
http://bugs.digium.com/view.php?id=12437. Thus far
I have created a branch for Asterisk 1.4 that changes the behavior
On Tue, Mar 10, 2009 at 12:19 PM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
dial-peer voice 5 voip
description ** **
preference 1
destination-pattern 1…
voice-class codec 1
session protocol sipv2
session target ipv4:1.1.1.1
session transport udp
dtmf-relay rtp-nte
fax-relay ecm
On Wed, Mar 11, 2009 at 7:32 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
I finally solved the issue by changing the resolution and the width of the
TIFF file to one that is accepted by the fax standard. In my case I changed
to a resolution of 96x96 and a width of 1728.
Now I am able
On Fri, Mar 13, 2009 at 11:38 AM, Marshall Henderson
marshall...@gmail.com wrote:
I recently read the thread entitled Faxing Success Rate on PRI which dealt
with Asterisk/HylaFax/IAXmodem. I'm successfully using this 'recipe' in a
few instances on systems with only a couple of analog lines all
On Fri, Mar 13, 2009 at 2:30 PM, Marshall Henderson
marshall...@gmail.com wrote:
On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg dbackeb...@gmail.com
wrote:
Again, you'll find people arguing that their voip solution has as low
of a failure rate as a hardware solution. I'm jealous. My voip
On Sat, Mar 14, 2009 at 12:00 AM, Steve Underwood ste...@coppice.org wrote:
Fully open-to-the-public FAX servers tend to get just get a lot of bad
calls, many of them wrong numbers, or voice users. FAX servers for
I've definitely seen that, and have been able to either identify the
validity of
On Sun, Mar 15, 2009 at 9:57 PM, MaxGao ss...@126.com wrote:
and many times when reciving tax , the E1 card will down , all the channel
get red alarm...
[Mar 16 09:49:19] DEBUG[20928] chan_dahdi.c: Monitor doohicky got event
Alarm on channel 2
[Mar 16 09:49:19] WARNING[20928] chan_dahdi.c:
On Mon, Mar 16, 2009 at 6:05 AM, dubravko caric
dubravko_ca...@yahoo.com wrote:
fax - SIP ATA (T38 enabled) - Asterisk #1 - IAX TRUNK - Asterisk #2 -
SIP ATA (T38 enabled) - fax
My question is, how can I know if I'm really using T38? is T38 information
coming to the other side (because of SIP
On Mon, Mar 16, 2009 at 11:29 AM, David Backeberg dbackeb...@gmail.com wrote:
On Mon, Mar 16, 2009 at 6:05 AM, dubravko caric
dubravko_ca...@yahoo.com wrote:
fax - SIP ATA (T38 enabled) - Asterisk #1 - IAX TRUNK - Asterisk #2 -
SIP ATA (T38 enabled) - fax
My question is, how can I know
On Tue, Mar 17, 2009 at 10:34 AM, Steve Davies davies...@gmail.com wrote:
While we have your attention Steve (Underwood) do you have a
high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We
currently use 0.0.4 with a very high success rate. Is there any
benefit in moving up to
On Mon, Mar 16, 2009 at 5:34 PM, Vincent Li vincent.mc...@gmail.com wrote:
Hello,
I just had a meeting about a pilot project going on in our University, The
project manager has done some research in the past year and concluded that
Asterisk can not scale well to large user base like 10,000
On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno tipas...@gmail.com wrote:
I have a weird problem with call using my T1 card. I can make calls fine
using my analog and IP phones, but when I try to initiate a call using a
.call file, I get the following error
-- Attempting call on
On Fri, Mar 20, 2009 at 5:36 AM, michel freiha mich...@gmail.com wrote:
Can you please help me in order to find the real issue?
Try taking out three or four pieces of your architecture, and then try again.
How about PSTN - Asterisk?
___
-- Bandwidth
On Tue, Mar 24, 2009 at 9:21 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
WARNING[12229]: app_fax.c:650 in transmit: Transmission error
and the ReceiveFax function ends abruptly.
That doesn't really help, other than that it seems your arrangement
defaulted to voice rather than using
On Tue, Mar 24, 2009 at 11:33 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
Sorry about that, I forgot to post them:
That all looks pretty good.
So in your original post, you clipped it off before you got all the
useful no-op output at the end.
I'm also assuming your file was empty?
On Thu, Mar 26, 2009 at 3:06 PM, Mike l...@virtutel.ca wrote:
Hi,
I know somebody is going to give me the link to the wiki hardware pages, but
I can't find the answer there. I'd like to know if, for an Asterisk only
system (nothing else of note running on it), I get a real gain from having 2
On Tue, Mar 24, 2009 at 1:57 PM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
Hello,
The NoOp output was not displayed at all. I'm assuming because of the
failure in the ReceiveFax application. In fact, the verbose output
Try changing
[fax-in]
exten =
On Wed, Apr 1, 2009 at 3:18 AM, Olle E. Johansson o...@edvina.net wrote:
What a shame about the loss of chan_hype. I was really hoping to build
a .com around it.
At least I'm feeling better since starting the placebo treatment for
my allergies.
___
--
On Thu, Apr 2, 2009 at 12:07 PM, Timothy Smith timotsm...@gmail.com wrote:
In our office, we're migrating from a Cisco set up to Asterisk.
What is the goal of doing this migration?
Plenty of people do a blended environment with Cisco doing what Cisco
does well and Asterisk doing what Asterisk
On Sat, Apr 4, 2009 at 8:31 AM, Roland Roland r_o_l_a_...@hotmail.com wrote:
I started with a digium card with ZAP
though that didn’t work out as the card were flawed..
so not to add more to my email, I'm seeking advice about the proper way to
learn about asterisk from A to Z if possible...
On Sat, Apr 4, 2009 at 11:18 AM, Timothy Smith timotsm...@gmail.com wrote:
We're migrating from Cisco to asterisk because cisco is expensive to
maintain, besides we can achieve more with asterisk like customised
IVRs etc.
I don't know what expensive to maintain means. We spend more on our
Hello there:
I think I have a silly kernel configuration problem. I'm running:
* vanilla 2.6.27.10 kernel built from source
* dahdi-2.1.0.4 built from source
So far so good,
dahdi module loads just fine:
dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.1.0.4
when I try to:
On Tue, Apr 7, 2009 at 11:54 AM, Gabriel - IP Guys
gabr...@impactteachers.com wrote:
I have a asterisk setup that is currently running on version 1.4.15 – I wish
to upgrade or migrate this instance to the current asterisk stable,
1.6.0.6. It is my intention to build a FC8 box, then install
On Thu, Apr 9, 2009 at 4:59 PM, John Rogers j...@wizworks.net wrote:
When I dial the extension of a meetme conference room, I get a message that
states is not a valid conference. The meetme app was working before.
I am getting this error on the CLI:
app_meetme.c:800 build_conf: Unable to
On Tue, Apr 14, 2009 at 7:39 AM, Tim Dobson li...@tdobson.net wrote:
I'm trying to convert some call recordings from asterisk we have in .gsm
Why not use sox for this purpose?
sox mygsm.gsm -r 8000 -c 1 mywave.wav resample -ql
Once it's a wav you can mp3 it with lame or your preferred encoder,
On Tue, Apr 14, 2009 at 7:11 AM, Michael mich...@networkstuff.co.nz wrote:
On Tue, 14 Apr 2009 23:00:02 Florian Hackenberger wrote:
Can somone spot the problem? Is someone using t38modem with asterisk
successfully?
The best advice I can offer is to give up now and use Callweaver otherwise you
On Tue, Apr 14, 2009 at 9:52 AM, Florian Hackenberger
f.hackenber...@chello.at wrote:
With asterisk 1.6, is it possible to use hylafax, or would asterisk
terminate the fax calls itself?
With app_fax integrated into asterisk-1.6, you have an 'infinite'
modem pool that you control through the
On Tue, Apr 14, 2009 at 10:46 AM, Lee Howard fax...@howardsilvan.com wrote:
David Backeberg wrote:
It may be possible to use hylafax, but
I don't know how or why you would.
The reason *why* is generally due to support issues.
What I was specifically getting at in the context of that response
On Wed, Apr 15, 2009 at 3:29 AM, Florian Hackenberger
f.hackenber...@chello.at wrote:
Thanks for the explanation! Sounds all good. There is one remaining
question however. As you mentioned T.30, is app_fax capable of
terminating T.38?
Yes although I'm speaking about 1.6. I can't say for
On Thu, Apr 16, 2009 at 10:08 AM, Massimiliano Stucchi
stuc...@willystudios.com wrote:
firmware. The problem arises when transferring a call coming in from a
SIP account to another phone. The call connects, but for the first 10
seconds there is a situation with one-way audio, then it turns
On Fri, Apr 17, 2009 at 2:47 PM, Jimmy Ezell jez...@hmhca.com wrote:
Our small company is replacing Cisco CallManager with Asterisk (because we
are tired of sending them money) and I am documenting the process as I go on
my blog. I am trying to make the notes as easy as possible in hopes
On Sat, Apr 18, 2009 at 10:27 AM, Thomas Kenyon
dig...@sanguinarius.co.uk wrote:
I will try this later, it looks straight forward enough. Does Asterisk
1.6 SendFax command autonegotiate T.38 (in the way callweaver does)?
Yes. Of course assuming that it's talking to a device that is T.38
On Mon, Apr 20, 2009 at 3:47 AM, Michael mich...@networkstuff.co.nz wrote:
Can this be used in the same way as Callweaver works, IE: to invoke Sendfax by
placing (using mv command) a job description file in it?
Yes. Callfiles can be used to initiate faxes.
On Mon, Apr 20, 2009 at 5:34 AM, Michael mich...@networkstuff.co.nz wrote:
Fax over T38 is failing, on the same system it worked with Callweaver.
Some people claim great success with callweaver. If Callweaver is
working great for you, why change what works?
What do I need to post to be get
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