Ok... I have heard this on Digium's PBX in the past, but can't seem to find
it anymore.
There was an IVR that you could dial into and Allison had recorded one of
the funniest messages I have ever heard... you are not the next caller,
hang up not, spend time with your children... it was
I have a sangoma 104d that is our main pbx now( legacy system died ). I
have replaced every phone in the building and things are going very well.
We have fax working well and calls are routing properly... All is well...
Except for our support modems... we have support people that dial out with
Steve,
I have the exact same problem with a sangoma a104d so I don't think it is
related to the card. I am trying to figure that one out as well,
fortunately I only have one DID on that trunk so I used _.* to route
everything...
Sorry this doesn't help other than to let you know that it
I have a system right now that has 32 extensions that I am setting up
hints for
snip
exten = 4521,hint,SIP/4521
exten = 4522,hint,SIP/4522
exten = 4523,hint,SIP/4523
exten = 4524,hint,SIP/4524
exten = 4525,hint,SIP/4525
/snip
The problem that I am running into is when I issue a reload, it
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We are looking for an asterisk guru / linux geek for full time
employment in the South West Virigina area. Please email me off list
for more details.
Sean
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Philippe Lindheimer wrote:
I would love to see some feedback on this as well. I've lost exact
count now, but think I've seen about 5-6 failures on their cards
TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I
don't deal with that
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It would not be the iaxy... it would be the phone that is attached to
it... there are plenty of phones/answering machines /other FXS
signalling devices that can do auto answer... the iaxy is not capable
of doing that...
Sean
Jerry Geis wrote:
Can
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Or a TDM2400 with 4 FXO modules... (4x4=16) :)
Lito Lampitoc wrote:
oh sorry, 2 TDM400P with 4 FXO modules each :=)
On 6/28/06, *Lito Lampitoc* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
or TDM400P with four FXO modules perhaps?
On
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What kind of T1? TDM? Data? What type of signaling are you planning
to use em? There is a lot of information that that question is
lacking for anyone to advise you ...
Jonathan Miller wrote:
I have a true leased line (a T1) between the two
. Being that I control
the termination at each end, do I get to specify the encoding?
On Wednesday 28 June 2006 10:17, Sean Cook wrote:
What kind of T1? TDM? Data? What type of signaling are you
planning
to use em? There is a lot of information that that question
is lacking for anyone
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It always helps to read the original post... so I apologize. I think
what you are looking to do is route the calls over the existing data
t1 in which case all you need to do is create an IAX trunk between the
two asterisk servers addressing their
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The only issues you could potentially run into is if all the modules
are FXS and they all needed to ring simultaneously... your power
supply may not be suited to handle to voltage requirements.
Sean
Ninneman, Tj wrote:
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Anyone have a 1650 running successfully in production mode with 2-4
PRI's? I want to make sure I don't have a motherboard compatibility
problem before I buy one of these. We are going to be using a Digium
TE210P to start off with and probably
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Double check to make sure you are actually running 1.6.6. I have it
working with 14 extensions right now with no problems...
Sean
Douglas Garstang wrote:
All,
Slightly off topic.
Polycom released their SIP software version 1.6.6 for their
Hi,
Anyone using SpanDSP with Digium TDM o TE cards to receive and email
Faxes?
No. Nobody ever uses this stuff. I just write it to waste my spare time.
Steve
Finish the coffee BEFORE checking the email...
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Off course for the price, you could by a four port sangoma with echo cancel
David Waugh wrote:
Hello
Eicon Networks produce a Single T1 card with hardware echo cancellation.
http://www.eicon.com/worldwide/products/MediaGateways/diva-server-vt1-pr
i.htm
-Original Message-
From:
I have been playing around with sipX for a couple of days now, and while
I don't really like it (just feels wierd), I do really like the
management interface for provisioning phones. I was wondering if anyone
had considered ripping this out of sipX or porting it to a simple php
interface or
Damon Estep wrote:
What you are proposing is quiet simple, and is done regularly. We
provision Linksys Sipura ATAs via a perl script with SSL and client
certificate authentication, as well as Polycom phones via XML file
drops. The newest Polycom firmware also states that ssl is supported,
but
Ok... I am reluctant to ask this question as I believe that it may be
like asking what someones favorite linux distribution is... but I need
to make an informed decision.
We are getting ready to upgrade from a TE210P to a quad T1 card with
echo cancellation. I am trying to decide between the
One of the primary differences between the two cards is the Sangoma
h/w echo canceler handles more cases of echo then do the Digium cards.
Whether you need that additional coverage is 100% dependent on your
specific implementation (eg, your T1/PRI provider), and not on what
the list thinks
hmmm... I am a huge fan of syslog-ng, but the stock syslog on your *
system should work well...
Matthew Warren wrote:
Does anyone know a good syslog server to use for grandstream phones? I want
to set this up to see what is happening with the grandstreams. Easy and
Free preferably.
syslog is limited to the amount of space you have on disk... I have
everything centralized by host/date on my syslog server.
In asterisk you can enable syslog by editing the logger.conf
;syslog keyword : This special keyword logs to syslog facility
;syslog.local0 = notice,warning,error
On the
Yes... it is very easy to do...
; on box a
exten = _NXXNXX,1,DIal(IAX2/boxb/${EXTEN})
;on box b
exten = _NXXNXX,1,Dial(IAX2/boxa/${EXTEN})
you just need to make sure that the context on the each side will have a
match for passing in ${EXTEN} to the other side
[from-boxa]
exten =
Just make something like this:
exten = 8863959,1,Macro(dial,8863959)
[macro-dial]
exten = s,1,Dial(SIP/${ARG1},60,r)
exten = s,2,NoOp(${DIALSTATUS})
exten = s,3,Voicemail,[EMAIL PROTECTED]
exten = s,104,Voicemail,b$([EMAIL PROTECTED]
exten = s,105,hangup
Just to make a bit more
Sean,
Where did you find that quote, I would like to see the rest of the
thread if there was relevant discussions.
Thanks.
It was really a two email thread... I had sent an email asking what the
status of BLA/SCA: Here is the entire thread:
Sean Cook wrote:
I take it SCA/BLA isn't
I would think the biggest issue with this at this point is because
Asterisk is not a SIP only platform, if one were to implement shared
line appearance, it would need to be designed in such a way that
channels other than SIP channels could participate in the sharing of
lines.
It has been
Is there a way through AMI or AGI to determine whether a channel is on
hold? Or if a channel has a call on hold?
Sean
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Actually... it means not in the production release. the subversion
trunk is a release but it is not for the faint at heart. While
generally everything works pretty well, it is expected that you will
find bugs and have issues :)
Sean
Douglas Garstang wrote:
In non-developer-speak, that means,
No, SVN trunk is not a 'release'. It's a development area. A 'release'
involves packaging it, documenting the changes, and handling bug reports
against it in a different way.
I guess I see release as a verb and not a noun. Probably is a good idea
to use the appropriate terminology... my
I take it SCA/BLA isn't going to make it into 1.4. Anyone have any idea
when support will be added to asterisk for this?
Sean
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OK... maybe I got a little anxious and ran out and bought a Tyan GX28
with dual Opteron (dual core) processors. (It is a nice server ;) ) I
did neglect to find out that you can not manually set the IRQ's on this
motherboard. I am now stuck sharing an IRQ with the ethernet
controller and no
Rob Lith wrote:
Does the sangoma handle sharing interrupts in some other way?
from: http://www.voip-info.org/wiki/view/Sangoma
There are no known compatibility issues (IRQ, IO etc) with ANY Sangoma
hardware and ANY make/brand of PC/server- NONE
___
Andrew Kohlsmith wrote:
On Thursday 25 May 2006 13:06, Sean Cook wrote:
There are no known compatibility issues (IRQ, IO etc) with ANY Sangoma
hardware and ANY make/brand of PC/server- NONE
Andrew,
Thank you for all of the information... I will clarify the any, none
verbiage
Kevin P. Fleming wrote:
Sean Cook wrote:
lspci -vb # shows IRQ 9 being shared
Interesting... I learned it from kenny in training in Huntsville ;)
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Kevin P. Fleming wrote:
Sean Cook wrote:
lspci -vb # shows IRQ 9 being shared
That is not a valid piece of information, and wherever you learned it
you should unlearn it :-)
'lspci -vb' shows the interrupt that the BIOS assigned to the PCI
device, but not where the kernel
a very elegant solution would be to do a Realtime lookup and match the
variable that is set.
exten = s,1,Answer()
exten = s,2,Realtime(sometag,cid,${CALLERID(num)},check_)
exten = s,3,gotoif($[${check_daughters}foo = foo]?...)
Then you can just add a record to the database...
On Wed, 2006-05-24
Not necessarily... my understanding is that the feature freeze was done
about 2 months ago for 1.4 and the release cycle is 6 months putting 1.4
due for release here pretty soon...
As to what is in the new release... probably most of the sip changes
that Olle has been working on as well as the
try running realtime load sip_buddies exten and see what you get...
SEan
On Thu, 2006-05-25 at 05:10 +0530, Chandan Mishra wrote:
Hi
i am using the asterisk server on one machine and mysql on another
machine.I have my mysql running on 192.168.77.75 and asterisk running
on the
Why not use exten = a,... and do a for more options press *, then
have it drop into an IVR...
Sean
Matt wrote:
Hi,
Is there anyway to add an option to dial someone from voicemail?
I know I can make 0 go to operator... however, I want to do something
like our Nortel did which was Press 7 to
Matt wrote:
That would be fine... and I know you can do alot of stuff with
Asterisk, you just have to think outside the box(tm) sometimes.
but my question with doing that is, then how do I make it go to this
message only when the person is not available, and not everytime
someone gets
Olle,
Is there a poster of you that I can put up on my wall ;)
Regards,
Sean
Olle E Johansson wrote:
Friends,
To update you on recent changes in svn trunk, I can inform you that we
now have ever smarter
ways to handle media streams in Asterisk than we do in 1.2 for the
IAX2 and SIP
have you tested to make sure that you can connect to the odbc resource
outside of asterisk via perl/php/(insert random language here)? Make
sure odbc is setup correctly and working before proceeding with the
asterisk part.
Sean
Dumpolid Exeplish wrote:
Hi,
I have duetifully followed the
I am running unixODBC to connect to postgres for your realtime data for
things like call forwarding, dnd and have noticed a significant delay
when running the realtime application. Has anyone else encountered this?
Even from the CLI if I do realtime load cf_data exten 4501 it lags for
[EMAIL PROTECTED] wrote:
Hello,
Digium does not provide snmp support to monitor their
cards !
That's like saying Toyota doesn't provide gas with their cars. You can
setup snmp with in linux and have it execute commands that you want to
determine whether or not the hardware is functioning
hm... why not just use ztdummy and save the $150 for the card?
Sean
Steve Totaro wrote:
I have a TDM4xxp card with no modules. My question is, will this card
be sufficient to provide timing or does it need to have modules?
Thanks,
Steve
___
See here is the interesting thing... zaptel alone is not sufficient to
provide timing... you need wctdm or something else loaded. But with out
modules, I don't think wctdm will load. (never tried it). Personally I
would spend $100 bucks and get a FXS or FXO module for it.
As for the X100P
(never tried it). Personally I
would spend $100 bucks and get a FXS or FXO module for it.
As for the X100P clone, the timing on them IMO is not anywhere near what
the tdm400. There are a few documented instance of timing just flat not
working.
Not working or irregular?
Our current PBX allows us to put a call on hold and then anyone in the
building can dial #9XXX and pick up that call. I know that I can
replicate a similar function by parking. But I would really like to
replicate the existing setup. Something about having to train people to
hit more than
Kevin Savoy wrote:
Can anyone recommend a phone to use in an inbound call center
environment that has an auto answer feature? We don’t want the agents
having to acknowledge the call. The call should just activate on the
headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601.
Try this one:
http://www.freedomphones.net/polycom/files/polycom.phone1cfg.pl-script
Sean
Andrew Kohlsmith wrote:
On Thursday 04 May 2006 14:45, Bruce Reeves wrote:
I am getting read to roll out close to 100 polycom phones and wondered if
any one knows of a program to take a list of MAC
sip.cfg
volume voice.volume.persist.handset=1
voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/
Jim Freeze wrote:
We are using the polycom 501 phones, and are having some challenges
with the volume setting. When a phone call comes in, the user ups the
volume for the handset,
couple of things... was asterisk compiled after zaptel? from the cli
try load chan_zap.so and see what you get
Ben Gore wrote:
This has got to be a stupid error I'm making...
I have been experimenting with different hardware and software
configurations before I decide what to use as a
my guess is that you are trying to dial a sip channel to reach an iax peer.
Dial(SIP/19)
should be
Dial(IAX2/19)
Olivier Saulnier wrote:
Hello,
I have some problems with a new configuration:
I always have on my asterisk console the message:
chan_iax2.c:5886 update registry: restricting
Haven't see this posted yet but keep in mind the polycom does offsets in
seconds not in hours... I spent three days figuring that out...
tcpIpApp.sntp.gmtOffset=-18000 is the same as GMT -5
Sean
Aaron Daniel wrote:
Whoops... meant dhcp... Keep in mind that if you're using windows' dns
Ok... I am not a telephone guy... I was born after rotary phones, so
forgive my ignorance in this matter. I am trying to get a really old
rotary phone up and running with an ATA. Why? Who knows... just
thought it would be cool. The problem is that it does not have an RJ11
connector, instead
Jerry Jones wrote:
Yellow=ground - not used
Green = tip
Red = ring
connect green/red to rj pins 4/5
You could pick up a quarter mod line cord (mod to spade) and replace
the cord, or use a screw terminal block to connect to line.
Enjoy
This worked perfectly! Thank you!
Sean
I do have a TDM400 and the Sangoma A200. I have done pulse with the
TDM400, but have not with the A200. I have just never seen a phone like
this... ;)
Rusty Dekema wrote:
On 4/25/06, Sean Cook [EMAIL PROTECTED] wrote:
This worked perfectly! Thank you!
Sean
Now, I think
Well it works! The pulse detection is a little squirrelly, even with
the debounce changes to wctdm.c. I can't get an audible ring but it
does work.
Sean
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On Mon, 2006-04-24 at 17:20 -0400, Mike Garey wrote:
As far as I can tell, after discussing this matter with other asterisk
users in my area, my telco _does_ provide disconnect supervision.. It
seems that the problem is actually related to the Sangoma A200 card
I'm using, as two other people
Try specifing [EMAIL PROTECTED] I know their have been some changes
with the implicit defining of the voicemail groupsthat may have
something to do with it... I didn't have to do anything special for my
polycoms.
Sean
On Fri, 2006-04-21 at 06:17 -0400, Andrew Kohlsmith wrote:
On Friday 21
Shouldn't be too difficult... perl has some great payment modules:
check out Business::OnlinePayment
http://search.cpan.org/author/MOCK/Business-OnlinePayment-StoredTransaction-0.01/lib/Business/OnlinePayment/StoredTransaction/Unstore.pm
modules on CPAN
Joseph wrote:
Is there a way somehow
We have had this problem with the TDM400 and just about every thing we
have ever had... it isn't the card that is chopping off the first
digit. It is the fact that it picks up too quickly and starts to dial.
Change your dial to be Zap/g0/w${EXTEN} and see if that takes care of
the problem
Is the Cisco 7960 capable of monitoring other extensions (hint status)
with a SIP implementation? Seems like it could, just can't find any
info on it...
Sean
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Are you using chan_sccp for you cisco implementation?
Aaron Daniel wrote:
Sadly, no. The SIP firmware on the Cisco phones doesn't support
subscribing to other lines. I heard chan_sccp does though.. now
to figure out how.
Aaron
On Thu, 6 Apr 2006, Sean Cook wrote:
Is the Cisco 7960
hahah... I have run into that dozens of times... you can even pull on
the cord a bit and it seems tight... then you give it a little more push
and click... you spend about 5 minutes thinking geez I am an idiot...
Sean
Matthew T. O'Connor wrote:
I had a one way audio problem with my Polycom
Aaron,
Here is really all you need:
exten = 401,Hint,SIP/401
in the context that the watching phone is in...
Aaron Daniel wrote:
The polycoms have a buddy feature where you can watch a buddy. From
what I can tell, it sends a subscribe to the server, and only works if
you're hinting the
I started with the polycoms to me its man those cisco phones boot
fast :)
Aaron Daniel wrote:
I think I'm getting there slowly... I notice in your extension, you're
hinting SIP/2348. I'll see if that helps me a bit, this damn phone
takes freaking forever to reboot.
Aaron
On Mon, 3 Apr
Is anyone using * to provide voicemail to a definity system? I
understand with the new SMDI functionality in trunk that this will be
easier to provide some of the integration features.
Looking for some hints on the definity setup and anything on the SMDI
side. Anyone with a working
My understanding is that the SMDI is a serial interface that passes data
about the call to the system for voicemail and pass MWI info back to the
avaya. It is the definity side that I am clueless on...
C F wrote:
On 3/27/06, Sean Cook [EMAIL PROTECTED] wrote:
Is anyone using * to provide
Yeah... I am doing that one now with a merlin system...
Sean
C F wrote:
Well, I did it using DTMF tones on analog channels, it's on the wiki.
On 3/27/06, Sean Cook [EMAIL PROTECTED] wrote:
My understanding is that the SMDI is a serial interface that passes data
about the call
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Aaron,
I have this working quite well. Are you using FTP? or TFTP...
We are using FTP for about 40 phones and it works like a champ. For
each phone I have...
0004f2030925.cfg
APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=phone4710.cfg,
sip.cfg
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I am currently running asterisk 1.0.9 on a system with 2 TDM400P... I
have had fairly good success with it across the board... my only issue
is that I have monkeys who move stuff around and things get unplugged ;)
Jared Davison wrote:
I would
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if you have an zaptel card installed and working... try to do a load
app_meetme.so and see what happens... if it loads successfully... you
should be able to conference also check your modules.conf and make
sure you don't have noload=app_meetme.so
with Asterisk is one of the Oh cool! moments.
On 3/23/06, *Sean Cook* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Now, here is what I'm not sure of at this moment. For the time
being, is it possible to just pass the PRI through the Asterisk
to the Legend? Will there by any type of dialplans
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Do hints work in Realtime asterisk? not finding much on the list
archives or anywhere else for that matter... I have tried using -1
priority as mentioned once or twice but no joy
Thought?
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Is anyone using the Adit 600 with CMG g729 gateway? We are trying to
come up with a solution for 600+ FXS campus and it appears to have the
highest port density of anything out there...
Any other thoughts?
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I would venture to say that all ITSP suck... some just suck less...
It generally speaking boils down to that fact that internet
connectivity is never full reliable (from a consumer standpoint).
Sure if you want to cough up the money for a T1, you
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Ok... so I spent today getting realtime extensions working, which they
are (for the most part) and apart from forgetting to commit
transactions in postgres and trying to figure out why an extension
won't work, all is well.
The only problem that I am
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Now, here is what I'm not sure of at this moment. For the time
being, is it possible to just pass the PRI through the Asterisk to
the Legend? Will there by any type of dialplans or anything that
need to be created? Will it pass the DID
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Check to make sure your minimum message length is very short. You
should be able to view this in the full log.
Sean
Phil Freed wrote:
Asterisk 1.2, Fedora Core 4:
When I leave a voicemail message, it writes the necessary files to
the INBOX:
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This sounds like a digitmap issue... from your sip.cfg what is your
digitmap set to?
Sean
sdgesa gaeharth wrote:
I am using the latest firmware and bootrom and this is a problem with
all 12 polycom 501s that we have in the office. If I want to
file? In other words, does asterisk tell
the phone what extensions are available and then the polycoms change the
map themselves?
thanks
*/Sean Cook [EMAIL PROTECTED]/* wrote:
This sounds like a digitmap issue... from your sip.cfg what is your
digitmap set to?
Sean
sdgesa
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I am looking to trade for a new or used Sangoma Analog A200 card with
echo cancellation. I have finished my testing with the OpenSwitch
card and want to test with the sangoma. Anyone out there looking to
do the same?
Sean
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The channels are VPB/X
On Fri, 2006-03-10 at 17:35 -0500, Chuck Fletcher wrote:
Any guidance on how to get my openline4 to get recognized by [EMAIL PROTECTED]
I've got my vpb drivers running, but not sure how to add it as a trunk,
should it be via zap? or is there another way?
Thanks,
I am using the odbc set up with postgres right now and it works fine.
http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL
has most of the info to get you running. As for meetme, I took the
app_cbmysql stuff for webmeetme and rewrote it for postgres. I am still
testing it, but it
Yes you do need unixODBC before you compile asterisk. Once you have
installed unixODBC , asterisk will compile and offer you the following
modules:
cdr_odbc.so
res_config_odbc.so
res_odbc.so
res_odbc.conf and cdr_odbc.conf are the related config files...
Sean
On Thu, 2006-03-09 at 11:57
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I actually have this working... on a merlin legend R7
zapata.conf
; turn off caller id otherwise it hangs...
usecallerid=no
usecallingpres=no
callwaitingcallerid=no
; drop into the vm context
relaxdtmf=yes
context=from-vm
group = 4
signalling =
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If memory servers me correctly DigitTimeout and ResponseTimeout are
depricated...
try:
exten = s,13,Set(TIMEOUT(digit)=5)
exten = s,14,Set(TIMEOUT(response)=30)
Sean
Robert P. McKenzie wrote:
Hello all. I'm having a problem debugging an IVR
To add to the other post... aah or amp actually has a DB that contains
call waiting information. It may have the default setup such that call
waiting is disabled. You should be able to dial *70 and enable it.
Sean
On Tue, 2006-03-07 at 11:33 -0700, Rolf Brusletto wrote:
All - I've been
First things first... use the latest version... (that I know of)
http://www.fitawi.com/Asterisk/
second... which part are you having problems with? The web piece? or
the app_cbmysql?
For the app_cbmysql, I have found that the easiest way to work with it
is to incorperate it into
I haven't tried sip yet... been finishing voicemail, but the principal
is the same.
res_mysql.conf
[general]
dbhost = localhost
dbname = asterisk
dbuser = someuser
dbpass = somepass
dbport = 3306
dbsock = /var/run/mysqld/mysqld.sock
extconfig.conf
voicemail = mysql,asterisk,voicemail
; i would
In theory I would say I agree how ever in practice... I have a PBX
(Merlin Legend) that I am connected to via PRI (10 foot pre-fab'ed
cable) and I get intermittent echo on the voip side. There is nothing
in between * and the PBX...
sean
On Fri, 2006-03-03 at 13:42 -0600, Michael Sampson wrote:
But even the FXO - voip bridging is lacking... you basically dial in
and it answers and provides dial tone for you to dial out your VoIP
service.
It doesn't provide incoming pots termination except to the FXS port.
Sean
On Wed, 2006-03-01 at 01:46 -0800, [EMAIL PROTECTED] wrote:
On Tue, 28
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what about ARI, it gives web based access to the voicemail and is pretty
good at it... the default vmail.cgi is probably not the best as it has a
gaping security hole that allows anyone to listen to anyone elses
messages :)
Sean
Martin Joseph wrote:
Just to through another hat in the ring... I use madplay for mp3s...
[default]
mode=custom
directory=/var/lib/asterisk/mohmp3
application=/usr/bin/madplay -Q -o raw:- --mono -R 8000 -a -12
On Thu, 2006-02-23 at 15:23 -0600, Aaron Daniel wrote:
I'd suggest using the format_mp3 program that's
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I have actually modified AMP to store the mac address and auto build the
phone.cfg and 0004XXX.cfg files for ftp. I use the default
username and password for the phones, so litterally all you do is plug
them in...
I will put together a
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Is it possible to pickup a call that is on hold on another extension?
Does anyone have any magic they can share on this topic?
I am struggling to teach call parking at a local shop where we installed
*. It would simplify my life so much if they
Same setup with two TDM400 (8FXO) running for over a year.
On Tue, 2006-02-21 at 01:37 +0100, Thomas Artner wrote:
Am Tuesday 21 February 2006 00:24 schrieb Marc Archer:
Hi All,
Can someone give me a definite answer as to wether or not you can
reliably run multiple TDM400P's in the
I believe that Centrex is ISDN correct?
Sean
On Tue, 2006-02-21 at 04:55 -0800, Dovid Bender wrote:
I do not know a lot about centrex but I know that most
PBX's support POTS lines (usually for faxing). You can
have them switch over the lines that they send you to
pots and then you can plug
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Why do you have immediate set?
*immediate*: Normally (i.e. with immediate set to 'no', the default),
when you lift an FXS handset, the Zaptel driver provides you a
dialtone and listens for digits that you dial, passing them on to
Asterisk. Asterisk
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