Hi,I already try that, unfortunately with no success. I wonder what is wrong?Cheers
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Hi,
it is a PRI. With the old telephone system the extensions were transmitted.
Only replaced the telephone systems and whatever I do, only the central
dial-in number is transmitted.
kind regards
Sebastian
Tom Vile [EMAIL PROTECTED] wrote:
Are you allowed to set your callerid with your
It seems that your asterisk cannot transcoding from ulaw to g729 and
vice versa.
What is the output from 'show translation' ?
Did you allow both codecs in sip.conf ?
Regards,
Stevanus
Il Neofita wrote:
I have the license for G729, however I need to use a
different codec for the prepaid
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hardcoding IP in /etc/hosts works but it's not reliable if voip
provider's IP change.
Does someone already as a similar problem and resolved it?
I head similar problem, but I didn't solve it. If you find workable solution,
please mail
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi,
I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.
For now, I use Via EPIA PD6000. Pretty small and
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I'm and [EMAIL PROTECTED] user - been so now for almost a year.
Lately, I've upgraded to the latest greatest.. (which is built on 1.2.5)
and am unable to install oh323.
I don't like to bring bad news, but I'm unable to install h323
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Guyz!
Is there any known issue with ooh323 and g729? I am experiencing one side
voice okay from ooh323 extension to sip ext, but on reverse side voice
quality is very poor. Any thoughts?
I have one question. Are you using ooh323 from
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I never so this error.
I am using H323 with Asterisk 1.2.6 Any idea what can be the problem?
You are using ooh323 from asterisk-addons-1.2.1?
If so, you are experiencing the same but that I have. There is patch (which I
can't apply) or
Anyone come across embedded device with FXO built in?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tomislav Parcina
Sent: Monday, April 03, 2006 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: Building
Hi,
I would like to find out;
Anyone using Asterisk Business Edition?
If yes, could you give us a brief description of what type
of solution you use it for?
What made you to choose the Business Edition over the Open
Source?
What advantage/disadvantage you have found?
On Thursday, March 30, 2006 8:27 PM Chris Earle wrote:
Someone plase help
This is terrible. Been trying everything for weeks :-(
Have you contacted Junghanns directly? They do not read this list. Usually
they are very helpful in such issues. Phone them.
Mit freundlichen Grüßen
On Friday, March 31, 2006 3:52 PM Chris Earle wrote:
I'm wondering if I should be using zapHFC with my Junghanns card
instead of qozap?
Why would you want to do that? Sorry if I missed the start of the problem
but qozap is what you want to use with your Junghanns card (at least if you
want
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Above products will work with any commercially
available motherboard in the market. This we have
proven true because we have at least 5 different
servers that we have installed them to. And take note
of their on-board echo cancellers,
Hi David,
We have no plans of charging for this service. I don't know which people you
are referring to that is has been forced to paying now - perhaps you would
let me know?
I know it might sound unbelieveable to you but Easy PABX is absolutely free
to use.
Thorben
Dovid Bender [EMAIL
I'm a retard, i knew i had seen it before... thanks.
--
~Shaun
Doug Lytle [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Shaun Reitan wrote:
I dont have that option in my phone, this is software 8.2 (p003-08-02 if
i remember correctly)
That is the firmware that I am also
Actually the outgoing call is going out through a sip channel,
and perhaps I should say the two calls. I am making two sip calls with one
dial command in the second priority:
[incoming_sip_calls_from_pstn]
exten = 3058472194,1,Dial(SIP/1035,10,r);## To ring
on the sip extension for
I would be very suprised if Digium allows even a mention of Sangoma
cards on the asterisk.org site. Digium owns the Asterisk trademark
and the asterisk.org site. Sangoma is a direct competitor to Digium
and their primary source of income, sales of hardware telco interface
cards(both analog and
On 03/04/06, Douglas Garstang [EMAIL PROTECTED] wrote:
The 'sip show channels' and 'show channels' command aren't exactly easy to
interpret, especially if one of the numbers has pic codes and rate centers
inserted (the rest is truncated on the output), or you have a proxy involved
in the
Hi All,
In previous mail lists, people talked about a solution
to record large amount of simultaneous calls. And then it seems that RAM disk
solution was the best choice due to the I/O bottleneck of Hard disk (System). Please
find the previous discussion as follows:
Hello,
What happens a few weeks into this when you've fragmented the free
space on the drives by deleting files periodically and rewriting in
some places? The consistent performance of SATA drives goes down
dramatically when this happens, much more so than on a SCSI-based
drive system.
Also, you
Using Dial() for this is not correct, because the Asterisk Dial() command
is not just for dialing a number, it also then connects the dialing with the
dialed channel, which is not what you want.
I had a close look into the Asterisk application and I thought
app_senddtmf will help, but the app
Hi,
I can not write CDR records in a macro after I do a dial
command and the line is picked up.
This is my situation:
After a pickup with a dial I jump to a specific macro. (dial,
with the extension M)
In this macro I want to write the CDR record with the
command
Hi!
I'm always getting this error when echo cancellation should start.
== ISDN3: Setting up echo canceller (PLCI=0x203, function=1,
options=4, tail=64)
== ISDN3: Setting up DTMF detector (PLCI=0x203, flag=1)
-- ISDN3: Error setting up echo canceller (PLCI=0x203)
Apr 3 10:49:30
Hi,
it is a PRI. With the old telephone system the extensions were transmitted.
Only replaced the telephone systems and whatever I do, only the central
dial-in number is transmitted.
kind regards
Sebastian
Tom Vile [EMAIL PROTECTED] wrote:
Are you allowed to set your callerid with your
Giuseppe wrote:
I'm always getting this error when echo cancellation should start.
What does your /etc/asterisk/capi.conf look like?
Also, have you configured your Eicon correctly? You may need to enable
the Eicon web interface and check that each port is correct.
--
National Manager -
Hi,
I have three R2 installation on different carriers, all shows the same
inconsistency at varying
degree. But, on most test calls we made, it reaches T3. The worst part of
these, the carrier
claims that it's my R2 box that is not responding in time. Please, check the
attached file and
take
Thanks Avi Miller!
Here it is my capi.conf file
[general]
nationalprefix=039
;;internationalprefix=011
rxgain=1.0
txgain=2.0
alaw=yes ;
[ISDN3] ;this example interface gets name 'ISDN1' and may be any
ntmode=yes ;if isdn card operates in nt mode, set this to
yes when
On Sun, Apr 02, 2006 at 12:55:13PM -0500, Kevin P. Fleming wrote:
Steve Kennedy wrote:
Each channel needs TWO licenses, one for each way (I think).
Nope. The encoder/decoder licenses are counted separately, and each
license you purchase entitles you to one encoder and one decoder.
Hmm,
Here's a good link.
http://www.asteriskguru.com/tutorials/sendtext.html
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
___
--Bandwidth and
Giuseppe wrote:
ntmode=yes ;if isdn card operates in nt mode, set this to
This should be set to no -- you should be in TE mode.
echotail=64 ;echo cancel tail setting
bridge=yes ;native bridging (CAPI line interconnect) if
I don't have either of these
has anyone seen a bristuff version compatible to the actual *1.2.6/zaptel
1.2.5 ?
the actual bristuff-patches version 0.3.0 PRE 1l doesn't apply correctly
anymore...
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
Hi
i've started a callcenter with Asterisk 1.2.4 a month ago, now i will
know, do you suggest to mantain the production system in line with the
stable release?
is the update process safe and secure?
thanks
___
--Bandwidth and Colocation provided by
Am Montag, 3. April 2006 11.35 schrieb [EMAIL PROTECTED]:
has anyone seen a bristuff version compatible to the actual *1.2.6/zaptel
1.2.5 ?
the actual bristuff-patches version 0.3.0 PRE 1l doesn't apply correctly
anymore...
No, but I had the same problem. Somebody told me (on this list) that
Which version of divas driver do you use?
If it's the v2 driver (from melware.de or in-kernel 2.6), then you need to
set
echocancelold=yes
in the interfaces sections.
If you use the new driver v3 (Eicon source RPM),
echocancelold=yes may not be set.
Armin
On Mon, 3 Apr 2006, Giuseppe
Am Montag, 3. April 2006 11.35 schrieb [EMAIL PROTECTED]:
has anyone seen a bristuff version compatible to the actual
*1.2.6/zaptel
1.2.5 ?
the actual bristuff-patches version 0.3.0 PRE 1l doesn't apply
correctly
anymore...
No, but I had the same problem. Somebody told me (on this
Please note that I forwarded this email from one
account to annother. If there is a problem with the
format please let me know and I will email direct. The
example below is a working config on a friends machine
Dovid
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
On the cmd Dial page Search
Could anyone please shed some light on this problem for a newbie:
Running Asterisk @ Home 1.7
When a call comes in (Zap Clone 100 Card), the extensions ring but when
lifting the receiver on an IP phone (BudgeTone 100), I just hear a bit of
crackle sound in time with the ringing of the other
This is possibly a dumb question, but I've googled
around and poked through
the documentation and I'm a bit confused.
My initial experiment with Asterisk involves setting
it up in place of my old
dedicated answering machine. That means I've
still got a regular old phone
on the line
Hi, thanks to you all!
Finally echo cancellation works correctly on my card (eicon diva server
4/bri)
Armin, that fixed the problem, thanks again!
Giuseppe
Armin Schindler ha scritto:
Which version of divas driver do you use?
If it's the v2 driver (from melware.de or in-kernel 2.6), then
Hi!
Is it possible to transfer a call to an external phone instead of
transferring the call to internal phone?
(I'm sorry for my bad english, I hope you understand)
When, during a call, I digit #123, the call is transferred to internal
extension 123,
but if I digit #external_phone_number, it
On 4/3/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
as mISDN neighter support fax-protocol nor beeing able two work in NT-Mode
it's no alternative for bristuff :(
I guess http://www.visdn.org/ is an alternative. I haven't used it
yet, but will be looking into it for sure.. (use daily builds)
Hi!
Is it possible to transfer a call to an external
phone instead of
transferring the call to internal phone?
(I'm sorry for my bad english, I hope you
understand)
When, during a call, I digit #123, the call is
transferred to internal
extension 123,
but if I digit
If you noticed I wrote the words it's only a hunch.
And that hunch is based on what has happend before.
--- Tom Vile [EMAIL PROTECTED] wrote:
Actually you are wrong about $$$ Dovid, I did not
charge one penny for it.
On 4/2/06, Dovid Bender [EMAIL PROTECTED]
wrote:
My guess (it's only a
Why you are not using span=1,1,0,cas,hdb3 ?
Since the other end is the Telco, they will provide you the source
timing sync.
Try upgrade your sangoma drivers and make sure the TE_CLOCK line inside
/etc/wanpipe/wanpipe1.conf says NORMAL.
[]'s
MM
-Original Message-
From: Dennis Nacino
Hi,
Is there a way to do a ZapBarge, but where the
person doing the
barge-in would be able to talk to the agent only
(whispering)?
Thanks,
There was a request for such a feature, however digium
said that since it would take a lot to create it theyw
ould charge $7,000.00. There
Hi list,
i am playing around with asterisk manager interface
(and astriskjava) and i notice that the login is not
case sensitive.
so i can use
username: admin
secret: admin
---
# telnet localhost 5038
Trying 127.0.0.1...
Connected to
Hi Sam,
There's an Asterisk module for OpenWRT running on the WRT54G and similar
Linksys WiFi routers. Is that embedded enough? :)
--
Regards,
Hilton Travis Phone: +61 (0)7 3344 3889
(Brisbane, Australia) Phone: +61 (0)419 792 394
Manager, Quark IT
G'day Heidi,
You have to remember that Digium is a direct competitor to Sangoma and
they will take as much time as they can to post praise of their
competitor's hardware. This is where OpenPBX is gaining hweadway - a
truly open source PBX.
Oh, and I run Asterisk here, not OpenPBX (at this point
Douglas Garstang wrote:
The 'sip show channels' and 'show channels' command aren't exactly easy to
interpret, especially if one of the numbers has pic codes and rate centers
inserted (the rest is truncated on the output), or you have a proxy involved in
the call. Wish someone with some C
as mISDN neighter support fax-protocol nor beeing able two work in NT-Mode
it's no alternative for bristuff :(
I have a mISDN installation working in NT mode here...
(Adding as second card today to be able to use Asterisk as Internet PBX
betweek my ISDN Phone and the Swisscom NT+2ab.)
Ok, I
Hi Dennis,
Update to libmfcr2-0.0.3 pre9. I made a slip in pre8. Sorry.
Steve
Dennis Nacino wrote:
Hi,
I have three R2 installation on different carriers, all shows the same
inconsistency at varying
degree. But, on most test calls we made, it reaches T3. The worst part of
these, the
If everyone stopped buying Digium cards and switched to another vendor
then Asterisk development would slow down and probably be less
organized.
It is important to mention that Sangoma has contributed many code
patches and new features to Asterisk in the last few years,
essentially giving Digium
We are using several providers and I would like to know if and how many
concurrent calls you can place for a voipstunt account?
Has anybody tried multiple concurrent calls?
bye
Ronald Wiplinger
___
--Bandwidth and Colocation provided by
This depends on how many servers you have handling your call center, and
if you have time where you can afford down time if you don't have
redundant servers.
It's generally best to find a version that works, and only upgrade if the
new version has bug fixes or feature additions that you would
On 4/2/06, Heidi Mendoza [EMAIL PROTECTED] wrote:
Hello List!I wanted to share to everyone the following compatibleconnectivity products that my company installed in our
Asterisk based soft switch. I already sent these tothe Asterisk.org site many days ago but for somereason they still have to
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Its also very important to recognize that some of us have problems with
some of the digium products where the equivalent sangoma product
resolves that problem. If it were not for the sangoma products, asterisk
would not be installed in
I think I sort of solved the problem.
It is related to Voipstunt provider. I tried it today with another
provider and all is fine. Well, I get some warnings from G729 like:
Jun 26 09:28:45 NOTICE[8440]: frame.c:179 __ast_smoother_feed:
Dropping extra frame of G.729 since we already have a VAD
I've been using my Asterisk (At my house - 2 modem-type
fxos, and an assortment of SIP endpoints for phones) for about 5 weeks now, and
I've been really happy with it, but I'm still having an echo problem that I've
exhausted google with, and can't get straight...
I think I've determined that
On Mon, 2006-04-03 at 07:19 -0500, Brian Roy wrote:
On 4/2/06, Heidi Mendoza [EMAIL PROTECTED] wrote:
Hello List!
I wanted to share to everyone the following compatible
connectivity products that my company installed in our
Asterisk based soft
Yes, as long as the context that the phone transfering has an exten
declared for that number.
On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote:
Hi!
Is it possible to transfer a call to an external phone instead of
transferring the call to internal phone?
(I'm sorry for my bad english, I hope you
Hello everyone.
This is an other question from a relatively newbie.
I'd like to provide auto callback ability for my *. From my mobile I want to
be able to call a number on the * and have it call me back on my mobile. I
know how to generate a .call file from a script and I know how to call a
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, April 03, 2006 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call transfer to external phone number
Yes, as long as the context that the
Hello everyone.
Is it possible to do some very basic voice recognition from within
Asterisk's dialplan? What I'm aiming at is the ability to speak the digits I
want to dial from my mobile phone. Dialing digits on my mobile phone while
driving is not all that safe...
Thanks for any input,
Cosmin
is there a difference between 1.2.1 and 1.2.2 ?
On 4/3/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Guyz!
Is there any known issue with ooh323 and g729? I am experiencing one side
voice okay from ooh323 extension to sip ext, but
Ok.. see it... so now my question is which should I use?
Obviously a hold system using ulaw for hold files is going to use less
CPU, but is it more stable to have Asterisk playing the sound files?
Especially since it has to start a seperate stream for every on hold
person? Seems like in a
Pete Barnwell wrote:
On Mon, 2006-04-03 at 07:19 -0500, Brian Roy wrote:
On 4/2/06, Heidi Mendoza [EMAIL PROTECTED] wrote:
Hello List!
I wanted to share to everyone the following compatible
connectivity products that my company installed in our
Asterisk
Have you tried madplay? I used to use it instead of mpg123 for MOH.
Can't remember whether it does start a new process for every instance or
not.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 03 April 2006 15:03
To: Asterisk Users
Hi there
Sphinx does speech recognition:
http://www.voip-info.org/wiki-Sphinx
HTH,
Cristi
On 4/3/06, Cosmin Prund [EMAIL PROTECTED] wrote:
Hello everyone.
Is it possible to do some very basic voice recognition from within
Asterisk's dialplan? What I'm aiming at is the ability to speak the
traceroute can't necessarily tell you how good a connection is, but it can tell
you how bad it is.
seeing as ICMP is usually first to be dropped, a good traceroute can be
indicative of no congestion.
And a bad traceroute may be indicative of congestion.
--
--
Steven
I wanted to share to everyone the following compatible
connectivity products that my company installed in our
Asterisk based soft switch. I already sent these to
the Asterisk.org site many days ago but for some
reason they still have to post it.
musiconhold.conf file:
[default]
mode=files
directory=/var/lib/asterisk/mohmp3
random=yes
Can anyone explain perhaps why my music is not being randomized? I
keep getting the same file starting the the queue. My callers are
getting bored as they hear the same music every time :P (Yes I have
Great info.
show channels concise does not show up when you enter show channels ?, so
thanks for the post.
--
--
Steven
http://www.glimasoutheast.org
Peter Fern [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Douglas Garstang wrote:
The 'sip show channels' and 'show channels'
Scratch that! Apparently asterisk just needed to be restarted..
wonder why a reload didn't work?
On 4/3/06, Matt [EMAIL PROTECTED] wrote:
musiconhold.conf file:
[default]
mode=files
directory=/var/lib/asterisk/mohmp3
random=yes
Can anyone explain perhaps why my music is not being
Mpg123 was the only way back in 1.0 versions.
In asterisk 1.2, they´ve implemented a new format module for mp3
codec, making it possible to do native mp3 streaming, but the old
way is still ok.
When I first saw the native one, there were some limitations, like the
mp3 could not have any ID3 tag
Of the people in here that have hinting working with the polycom 601's (or
any phone for that matter)... do you have it working so that the shared
line appearance shows that there's someone on the phone? If so, any hints
on how to do it?
--
Aaron Daniel
Computer Systems Technician
Sam
Hi,
It seems as 'the google' has left me today so I am trying the list.
How do I get access to SIP responsecodes from dialplan/agi. Yes I know
that I should stay with 'DIALSTATUS' but there are cases where I need
the responsecode like '484 adress incomplete' and not just the 'NO
ANSWER'
Thanks!
Can't believe it actually exists...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Cristian Draghici
Sent: Monday, April 03, 2006 5:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Asteirsk has got no clue what's internal and what's not, it's the
context that decide what numbers are available for a user.
In your case more info is needed to troubleshoot it.
On 4/3/06, Cosmin Prund [EMAIL PROTECTED] wrote:
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]
Madplay did work fine with * 1.0.
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andre Ruiz
Sent: 03 April 2006 15:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?
Mpg123
Hi,
I am renovating my house completely and installing
new cabling for communication.
I'm not to into this PBX thing but I would like to
have a simple one for my house, to have different phone numbers for my family
members, some kind of integration with my Broadband telephone, and
Do you have an idea when this new submission will be available?Thanks. Dan Austin [EMAIL PROTECTED] wrote: Sorry for the late reply, I was away on vacation.Version 1.2 was created by Areski and I extended it to include the scheduling functions. I guess I should get an account on
Hi,
I am able to use SpanDSP 0.0.2 (all pre version) with Asterisk 1.2.6 to
recieve fax successfully. Today I tried to change to SpanDSP 0.0.3 pre 6
but I just couldn't complie the app_rxfax and txfax application. The
SpanDSP 0.0.3 was successfully complied though.
I'm running it on Debian.
Erik A TE110P would not really be
appropriate, unless you plan on having a T1 in your house. You could run
Asterisk @ Home on a dedicated box, and something like a TDM21B, which has (1)
FXO port and (2) FXS ports. You could run your existing analog (POTS) phone
line into the FXO port, and
Currently Asterisk will not integrate with Skype. You would
need a provider such as Teliax, Broadvoice, IAX.cc or many others or you can use
hardware devices to connect to traditional phone lines. You didn't say what
broadband phone you have but if its Vonage, there are also issues with
SIP transfers happen out of band, so the context is the sip phone's
context noted in sip.conf.
For Inbound and outbound (ie Dial application), the context is the entry
point in the dial plan. If you need features.conf transfers to work in
a specific context you need to set the __TRANSFER_CONTEXT
On Sat, 1 Apr 2006, Matt wrote:
However, anyone have a good way to log the agent out without having
them enter their agent ID and then have to hit # for the new
extension?
There are a couple of ways listed here in the Wiki:
On 3 Apr 2006, at 14:43, Cosmin Prund wrote:
Hello everyone.
This is an other question from a relatively newbie.
I'd like to provide auto callback ability for my *. From my mobile
I want to
be able to call a number on the * and have it call me back on my
mobile. I
know how to generate a
Hello,
I can use directed call pickup using pickup application (between sip,
iax, skinny cals),
but unable to pickup call that is ringing on phone behind h323 gateway
(using original h323 channel in asterisk), is this even suported?
thx
PJ
exten = _*7.,1,Pickup(${EXTEN:2})
console log,
Hi, im running asterisk as another user (user x group x), While trying
to attach to the console I get this error:
[EMAIL PROTECTED] keys]# asterisk -r
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
[EMAIL PROTECTED] keys]#
the file exists and it is:
srwxr-xr-x 1
It can't really do a whole heck of a lot though.
-Original Message-
From: Quark IT - Hilton Travis [mailto:[EMAIL PROTECTED]
Sent: Monday, April 03, 2006 4:37 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Building
Everyone is free to implement products based on what their heart tells them
and not their head. I don't know a lot of people that manage to stay in
business very long doing things that way though.
-Original Message-
From: Tomislav Parcina [mailto:[EMAIL PROTECTED]
Sent: Monday,
What about the Asterisk Developer pack, a good Linux box and his
standard phone line?? I've seen this work and it does a great job. The
Telco doesn't know anything as Asterisk integrates with the analog
phone line and things just work.
Am I off base here??
RandyW
Kerry Garrison wrote:
recieve fax successfully. Today I tried to change to SpanDSP 0.0.3 pre 6
but I just couldn't complie the app_rxfax and txfax application. The
SpanDSP 0.0.3 was successfully complied though.
.3 is for developers only it is not intended for enduser use.
Oh, sorry, I meant the pipe system (external program), not exactly
only mpg123. Madplay indeed did work fine with * 1.0.
On 4/3/06, Lee Archer [EMAIL PROTECTED] wrote:
Madplay did work fine with * 1.0.
Lee
___
--Bandwidth and Colocation provided by
the user you are connecting as should have full rights to /var/run/asterisk:
http://www.voip-info.org/wiki-Asterisk+non-root
hth
-Original Message-
From: Erick Perez [mailto:[EMAIL PROTECTED]
Sent: Monday, April 03, 2006 9:28 AM
To: Asterisk Users Mailing List - Non-Commercial
I've never had issue with the Digium cards in testing and as we're
looking forward to production systems what compelling reason do I have
to pick Sangoma? (I'm not looking for a flame-fest here, but actual
compelling reasons, ie Sangoma cards support foo which is needed in
situation bar and
Pavel Jezek wrote:
Hello,
I can use directed call pickup using pickup application (between sip,
iax, skinny cals),
but unable to pickup call that is ringing on phone behind h323 gateway
(using original h323 channel in asterisk), is this even suported?
thx
PJ
exten =
"or you can use hardware devices to connect to
traditional phone lines"
That can be a Digium card, Sangoma card, Linsys
SPA3000, Mediatrix 1204, and several other devices.
Kerry GarrisonDirector of
Technical ServicesTech Data Pros - Orange County's Mobile IT Service
Provider(949)502-7819
Douglas Garstang wrote:
The 'sip show channels' and 'show channels' command aren't exactly easy to
interpret, especially if one of the numbers has pic codes and rate centers
inserted (the rest is truncated on the output), or you have a proxy involved in
the call. Wish someone with some C
Aaron Daniel wrote:
Of the people in here that have hinting working with the polycom 601's
(or any phone for that matter)... do you have it working so that the
shared line appearance shows that there's someone on the phone? If so,
any hints on how to do it?
It's not a shared line appearance.
1 - 100 of 199 matches
Mail list logo
