Has anyone on this list gotten hold of these cards? It's been 2 months
since their official ship date.
Even the website www.ipvolution.com is in wee-wee land.
/leo
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On Fri, 11 Mar 2005 14:41:37 -0500, C F [EMAIL PROTECTED] wrote:
Welcome to SIP, this is how SIP works, thats why ppl use IAX.
It is a combination of chan_sip and the particular sip.conf actually.
Sane SIP servers will challenge all INVITEs, and apply user
identification from the user
Try using dtmfmode=rfc2833 in your sip.conf.
It should work...
Hope, this could help.
Guido Hecken
I have a question,
I am unclear on how to park a call. I know that you are supposed to be
able to press # and then transfer the call to extension 700. However,
* doesn't seem to be graping the
J Thomas wrote:
We have given a few PAP2-NA to our business customers with both phone
ports configured through the same SIP server. We cannot call them both
at the same time. Surprisingly, we can call both the phones one at a
time fine. Is there something we are missing in the configuration?
Any
I met a strange SIP problem recently.
In an ordinary procedure, when asterisk loads sip module, a series of
functions are called sequentially:
load_module()-restart_monitor()-ast_pthread_create()-pthread_create()-do_monitor()
However in my system, pthread_create() failed to create a child thread
Does anobody know an IAX2 software phone for
PocketPC?
Regards,
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Hi,
I am playing around with SIP extensions on my local lan using X-Lite but I
am having a bit of difficulty, I have set up X-Lite and my sip.conf
accordingly, but when I start it I get the following message:
Login failed! Contact Network Admin
I am still able to dial local extensions on my *
Hi,
Where can I find the code that performs the voice e-mail function (that
is, the code that reads the contents of voicemail.conf and then performs
the necessary action)?
I am using [EMAIL PROTECTED] 0.6.
Thanks in advance!
--
Rgds,
Julius.
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af kurt x
Sendt: 9. marts 2005 20:57
Til: Chris Wade
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [Asterisk-Users] GotoIf problem
I,ve gotten the GotoIf
Leo Ann Boon wrote:
Has anyone on this list gotten hold of these cards? It's been 2 months
since their official ship date.
Even the website www.ipvolution.com is in wee-wee land.
It has been down for several weeks. The cards are still shown on
www.atacomm.com. I don't know whether that is a
On Sat, 12 Mar 2005 13:03:00 +0300 (EAT), Julius Kidubuka
[EMAIL PROTECTED] wrote:
Hi,
Where can I find the code that performs the voice e-mail function (that
is, the code that reads the contents of voicemail.conf and then performs
the necessary action)?
I am using [EMAIL PROTECTED] 0.6.
What issues/options are there when forwarding voicemail to uk mobile
voicemail?
ta
Rafal Kaniewski
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.7.2 - Release Date: 11/03/2005
I have no idea. I live in the USA so I don't normally need busydetect.
Anton Krall wrote:
Why does busydetect actually drop calls while stile talking?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Viernes, 11 de Marzo de 2005 03:58
Hi there,
Can anyone help me with a problem I have setting up my VEGA 50 BRI gateway
with Asterisk? I have been successful making outgoung calls, but have been
unable to get the Vega to register with Asterisk.
Would anyone have a sample section of Sip.conf to help me? Does Asterisk
currently
On Sat, Mar 12, 2005 at 01:03:00PM +0300, Julius Kidubuka wrote:
Hi,
Where can I find the code that performs the voice e-mail function (that
is, the code that reads the contents of voicemail.conf and then performs
the necessary action)?
I am using [EMAIL PROTECTED] 0.6.
The mail is
Florian Overkamp wrote:
Hi,
-Original Message-
You should be able to download one (for WIndows and possibly Mac) from
efax or j2.com I think.
http://www.efax.com/en/efax/twa/page/download?rqcp=2
http://www.j2.com/jconnect/twa/page/download
You might be able to do that, but take a
I have seen the list of codecs for the ATA 186's but not sure if it was
100% or not.
I want to know really is it possible to run GSM or ilbc on them or is a
G729 lic the only way to get a low bandwidth codec?
This is the list of codecs that I have seen.
RxCodec and TxCodecConfigure the codec
[EMAIL PROTECTED] wrote:
I know this is a bit off topic but we are using Asterisk :)
Since this list is full of tech gurus w/ all different sorts of
backgrounds, I thought I would get the best opinions here.
We have several different switches and other telecom equipment at our
facilities which
Ok, I've seen this question go unanswered on the mailing list, and I assume
it's because no
one had the heart to break the bad
news to the guy asking, but be honest with me, I can take it. At this time
it's flat
impossible to have multiple IAX phones
behind a NAT without using an *
would like to know if some of you have tested asterisk connected to an
EADS 6550 analogique PBX (also know as Nexpan50).
Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no
other card, each of them have their own IRQ) all ports connected to the
EADS. We have GS ATA286
I have a strange situation. Once in a while (non-deterministic) the 2 TDM04B
cards lock up at the same time and stop processing incoming and outgoing
calls even though * shows that it is trying to communicate to ZAP channels
(at least on the outgoing). The only cure is to reboot the system
David Uzzell wrote:
I have seen the list of codecs for the ATA 186's but not sure if it was
100% or not.
I want to know really is it possible to run GSM or ilbc on them or is a
G729 lic the only way to get a low bandwidth codec?
This is the list of codecs that I have seen.
RxCodec and
Hi,
We did an interconnection with our carrier few days ago. But, I noticed
that there was a signaling problem on our trnuk. In fact, Asterisk indicates
that the call is answered when we received ALTERTING message from our
carrier. This is PRI debug logs :
-- Executing Dial(IAX2/[EMAIL
I think you missed out a *NOT* below...
In short, you *CANNOT* install or otherwise use any hardware cards, like
Zaptel, with Asterisk when running on CoLinux and generally, I'll advise
you to not use Astwind for anything other then playing. It's a nice toy,
but that is all.
-Original
Grett. This should be loads of fun then... 8(
I have noticed what I can only describe a negative undertone with
several VoIP poviders.
Not an easy customer? We don't want you. Things like that.
The LiveVoIP website is in fact like that.
There are several places on the site that
Hi,
Is there anyone with some uptodate info on the Cisco 7970 and Asterisk
skinny / SCCP?
I know chan_sccp doesn't support the 7970.
Is it true the basics should work with skinny?
All info is welcome.
Steven
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Guys, this is weird.. Today I started having some problems with calls been
dropped. Im suing X100p cards (clones) and I have this setting on my zatala
fle:
busydetect=yes
busycount=4
Try changing busycount to 6 or 8, stop asterisk, and restart.
How did you go?
On Tue, 8 Mar 2005 11:28:59 +1030, Peter Childs [EMAIL PROTECTED] wrote:
Digium shipped me a replacement card, but they sent the wrong one, so they
fedex'd another and its just arrived.
Should be testing in the next two days (the box is in another state...)
The last I
Rich Adamson a écrit :
would like to know if some of you have tested asterisk connected to an
EADS 6550 analogique PBX (also know as Nexpan50).
Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no
other card, each of them have their own IRQ) all ports connected to the
EADS.
I saw some coverage of this in the list archive but no
one seems to have
posted a resolution.
I am using [EMAIL PROTECTED] 0.06 and when I get a call from
LiveVoip over
IAX I dump it into my IVR.
From there the call is routed to groups based upon
input.
However, there is no
Hello People,
I have a Sipura SPA-2100 with default configuration and the last software
upgrade, and a * from Debian Sarge with the simple configuration:
[general]
port = 5060
bindaddr = 0.0.0.0
[103]
username=103
type=friend
secret=qaz123wsx
qualify=no
port=5060
nat=yes
host=dynamic
we have a full line of WiFi phone that work great. Works at any hot spot
also
for more info contact me at [EMAIL PROTECTED]
- Original Message -
From: Sylvain COUTANT [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, March 11, 2005 12:42 PM
Subject: [Asterisk-Users]
would like to know if some of you have tested asterisk connected to an
EADS 6550 analogique PBX (also know as Nexpan50).
Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no
other card, each of them have their own IRQ) all ports connected to the
EADS. We have GS ATA286
Hi all,
I'm a newbie and I have a configuration problem with Asterisk.
Seems that I'm not able to call an outbound number. I'm quite sure that it
is a configuration problem, but I'm not able to find out where is the
mistake, even reading several docs to www.voip-info.org.
I do not have a good
[200]
type=friend
username=richard
change this to
username=200
And my X-lite Default SIP Proxy config is as follows:
Enabled: Yes
Display name: richard
Username: richard
Change this to
Username: 200
and this one
Authorisation User: richard
to
Authorisation User: 200
and it should
Mark Matthew,
I know how frustrating it may be, ...
I can imagine your feelings, ...
HOWEVER, with all respect, it does not help me to fix my problem!
Can we come back to the subject, please?
I apologies for the missing words for me in the Subject!
I tried to follow (and may made some mistakes)
Stephen Misel wrote:
Ronald Wiplinger wrote:
New developments in our business plan make a change necessary.
We would like to offer different prices, depending on the
user/calling card. How can we use that with ASTCC? or should we use
something else?
ASTCC allows for multiple brands and
Rich Adamson a écrit :
would like to know if some of you have tested asterisk connected to an
EADS 6550 analogique PBX (also know as Nexpan50).
Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no
other card, each of them have their own IRQ) all ports connected to the
EADS.
On Fri, 11 Mar 2005 16:38:38 -0600, Nathan Bowyer [EMAIL PROTECTED] wrote:
On Fri, 11 Mar 2005 15:04:03 -0600, Matthew Boehm [EMAIL PROTECTED] wrote:
You can't have this:
[from-sip]
switch = Realtime/[EMAIL PROTECTED]
The context in your extensions.conf must be different from your
Hello All,
How are you all doing today? Good I hope.
Well, I have been learning a lot about Asterisk and must say that it is
GREAT
I have been able to configure and compile various ideas that I have been
working on in my test Asterisk PBX and everything worked just fine.
The reason for
So, should I just disable busydetect then?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Sábado, 12 de Marzo de 2005 04:13 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Droping calls
Will give it a try. Thx!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Sábado, 12 de Marzo de 2005 07:15 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Droping calls
Guys, this is
Are you running any NAT anywhere? I see that your NAT value is set to '1'.
It should be 'yes' or 'never'. That might be your problem.
Have you tried adding this user into the sip.conf first to verify that this
is truly an ARA problem?
-Matthew
From: Ronald Wiplinger [EMAIL PROTECTED]
I am trying to figure out a way to add something like:
61 100 pennies (Everything what is not
listed below)
61 78150
61 5 130
61 342 180
How could I do these (four) regex?
bye
Ronald
sipfriends = mysql,astconf,sip_buddies
Yes. Remove that line. This was done a few weeks ago to better split
peers/users.
sip show users and sip show peers does not show the phone, but
Go into sip.conf and enable the 3 RealTime cacheing variables. This will
make them show up in the
hi
having a queue with some SIP members, is there a way to check how many
of them are connected to asterisk, and if none are, go to a different
context?
thanks
roy
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On Sat, Mar 12, 2005 at 04:04:29PM +0100, Androtech wrote:
Hi all,
I'm a newbie and I have a configuration problem with Asterisk.
Seems that I'm not able to call an outbound number. I'm quite sure that it
is a configuration problem, but I'm not able to find out where is the
mistake, even
Roy Sigurd Karlsbakk wrote:
having a queue with some SIP members, is there a way to check how many
of them are connected to asterisk, and if none are, go to a different
context?
Not at the moment, no.
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Ronald Wiplinger wrote:
Stephen Misel wrote:
Ronald Wiplinger wrote:
New developments in our business plan make a change necessary.
We would like to offer different prices, depending on the
user/calling card. How can we use that with ASTCC? or should we use
something else?
ASTCC allows for
would like to know if some of you have tested asterisk connected to an
EADS 6550 analogique PBX (also know as Nexpan50).
Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no
other card, each of them have their own IRQ) all ports connected to the
EADS. We have GS ATA286
Stephen Misel wrote:
Ronald Wiplinger wrote:
New developments in our business plan make a change necessary.
We would like to offer different prices, depending on the
user/calling card. How can we use that with ASTCC? or should we use
something else?
ASTCC allows for multiple brands and
I do not undertand how to tell to Asterisk to use the X100P to dial an
external number instead the internal one. It always calls the internal
extension.
Could someone give me a valid iax.conf and extension.conf examples files.
Regards,
chan_iax2 says you have no section in your iax.conf
Hi,
I tried that but same error
Specially I didn't find people posting about Bad Request or Unknown Dialog
-- Got SIP response 400 Bad request back from 147.135.8.128
-- SIP/5092321848-ccd4 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing
It doesn't try to authenticate the incoming call.
On Friday 11 March 2005 03:56, Randy Johnson wrote:
What does insecure=very do?
Dan Weber wrote:
Here is my sip.conf:
===
register =
[EMAIL PROTECTED]::[EMAIL
Thanks,
I have that already in my /etc/hosts
But it's still not working :(
On Saturday 12 March 2005 03:48, Rich Adamson wrote:
For everyone that's trying to get BV to work, you'all might want to
edit your /etc/hosts file and insert something like:
147.135.8.128 sip.broadvoice.com
This
Hi,
I have been playing about with meetme as a conference bridge, and find it
lacking in some features which I believe are out their somewhere.
Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design
it looks like a plan happened, but where is meetme2 at now?
Things like
Hi,
my SIP provider sends me all the numbers that are dialed, i.e. when a
number is appended to the phone number proper, it gets appended to the
incoming number, like (my number is 0123456789, and I append a 5):
To: sip:[EMAIL PROTECTED];tag=as1174b008
Question is, how can I use the trailing
New PHP web interface at www.fitawi.com/Asterisk
There is also a sample cbmysql config file and the database
Tables description.
-New in the interface:
Added schedule conflict detection for Add and Update conference
functions.
-New in app_cbmysql:
No
Thanks for the reply. Here is my sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default
disallow=all
allow=ulaw
dtmfmode=rfc2833
register =
[EMAIL PROTECTED]:password:[EMAIL PROTECTED]/
canreinvite = no
Hello Justin,
dtmfmode should be inband for broadvoice either way because that's what they
support.
Now for the extensions.conf do you have:
exten = ,1,Dial([SIP|IAX2|..]/something, timeout, t) -- t for
transfers?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Don't sweat it - it just so happens that you came into the fray just moments
after this list has had a big long drawn out argument about newbie etiquete
(sp?). You've just managed to get caught in the middle. Don't let it be
indicative of how everyone feels, and don't let it scare you away...
administrator tootai wrote:
If you're telling that I have to pass parameters to module when loading,
I checked with modinfo wctdm (at office I have head version) and options
I have are those:
[EMAIL PROTECTED] asterisk]# /sbin/modinfo -p wctdm
debug int
loopcurrent int
robust int
_opermode int
On March 11, 2005 10:49 pm, M.N.A.Smadi wrote:
does any body know what are the physical dimension of a digium care
400pm for example?
It's a half-length PCI card. It's maybe an inch or so longer than the PCI
slot itself. It is full-height though (i.e. the circuit board is as tall as
the
Dear ALL:
Where can I find the oh323 module on CVS or anywhere?
I want to implement the SIP(ser) to Asterisk to H323(gnugk).
Thank you.
Best Regards
Charles
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On March 11, 2005 04:51 pm, Eric Wieling wrote:
echocancel=yes
echotraining=yes or 600 or 800
I absolutely *despise* echotraining. A half second (or in your case 8/10 of a
second) delay before hearing anything is unacceptable in almost all
situations.
Maybe if you've got a physical
On March 11, 2005 10:49 am, John Goerzen wrote:
* Then there's the NDA: People are specifically prohibited from
telling anyone that they use VoipJet, including end users. Also,
we can't tell people what we pay for it, even though the prices are
right there on their website.
*
On March 11, 2005 11:13 am, Ronald Wiplinger wrote:
I have ASTCC installed, and compare it with NuFone, however, I find that
the billing of NuFone is always a few secondes more (6 to 24 seconds)
Does anybody has an explanation / solution for it?
Have you emailled [EMAIL PROTECTED] and asked
NuFone service bills in industry standard billing increments, which
are: six (6) seconds for the US48, sixty (60) seconds to Mexico and
fifteen (15) seconds to the remainder of the world.
From: http://www.nufone.net/tac.html
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Asterisk-Users mailing
Title: RE: [Asterisk-Users] No ringback over IAX - LiveVoip
Excellent thing to
hear. I am glad there are positives on this site as well as teh
warnings.
Now to get the ringback issue
resolved
Using m switch to get MOH is OK but there
has to be alogical reason this is occuring adn a
On March 10, 2005 07:14 pm, Giudice, Salvatore wrote:
I vote for MySQL. PostgreSQL is fine, but MySQL handles much better
under extreme load. MySQL is also usually touted as being generally
You *gotta* be kidding me. MySQL can't hold a candle to PostgreSQL for high
load, high volume or
Dear list,
I am trying to learn how to use Zap-things in Asterisk.
While loading Asterisk verbosely I get this error:
[chan_zap.so]Warning, flexibel rate not heavily tested!
= (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Mar 12 17:19:01 WARNING[5563]:
Hello,
I found a way to do a callback service using call files however, only
call leg B gets recorded in the CDR so I would only be able to
accurately bill for one leg?
Does anyone have any suggestions on how to get numbers for both call
legs? Or a way to bridge two seperate SIP calls and have
Hi All,
I've been trying for a while to get * to play MusicOnHold with my SIP
connection. I can hear it when I call a test extension from my local X-Lite
phone, but when I dial in via InterVivo, I just hear silence.
I have a Gentoo box with kernel 2.4.28-gentoo. I have no sound card or
On Sat, 12 Mar 2005, C. Tomlinson wrote:
I have been playing about with meetme as a conference bridge, and find it
lacking in some features which I believe are out their somewhere.
Viewing this wiki page http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design
it looks like a plan happened,
On Sat, 12 Mar 2005, Laurent Tostain wrote:
Hi,
We did an interconnection with our carrier few days ago. But, I noticed
that there was a signaling problem on our trnuk. In fact, Asterisk indicates
that the call is answered when we received ALTERTING message from our
carrier. This is
Hi
Can u let me know how you are doing this, please
Iqbal
On 3/12/2005, Rafal Kaniewski [EMAIL PROTECTED] wrote:
What issues/options are there when forwarding voicemail to uk mobile
voicemail?
ta
Rafal Kaniewski
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Hello All,
I'm just getting into *, and trying to use a Broadvoice account. It
works inbound, but Outbound fails no matter what sip.conf parameters I try.
From the recent posts here I think it could be:
A bad CVS release - I will try to download and build from a new one
Broadvoice
Pulver.communicator (FWD) ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
FCG ZHAO Zigang
Sent: Freitag, 11. Mrz 2005 06:17
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] what is best free softphone.
I use xlite , but it
On Fri, Mar 11, 2005 at 01:56:47PM -0500, Giudice, Salvatore wrote:
As for the production recommendation you refer to, I would
respectufully disagree. If you are an enterprise comapny looking to
deploy an open-source DB, you will pick the one that has an established
support company to
... I just tried again after removing my hosts file entry (again) and
outbound is now working! I had taken it out before, but I think I was
getting a different error at the time.
Sometimes it seems like asking for help is itself a cure!
Thanks anyway!
JDC
I am working for Accudata Technologies. We
provide CNAM via http request or raw TCP/IP connection.
We would like to provide the same capability to Asterisk. I installed
Asterisk on Fedora 2.0
and did reading about AGI and AGI application at
Hi All;
I know Asterisk can support video calls over sip or
h323 but I need to know if it can be used in Video Conferencing?
Can I use "meet me" for that purpose?
Regards
Mohammad
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On Fri, Mar 11, 2005 at 04:25:59PM -0500, David Filion wrote:
Maybe I miss read, but doesn't the licensing of the newer releases of
MySQL require companies to purchase a license?
No. The license is GPL. Originally it was LGPL for the client libraries
but this got changed recently.
So you
Kevin Nguyen wrote:
Thank you for your help.
Kevin N.
I already replied to your first message with a great deal of
information; did you not receive it?
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After upgrading to asterisk 1.0.6 on Gentoo when I try to log-in to
check the voice mail I get:
app_voicemail.c:3389 vm_execmain: Unable to read password
--
#Joseph
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These allow and disallow work with NuFone for me
disallow=all
allow=ulaw
allow=alaw
allow=gsm
Jeff
Message: 11
Date: Fri, 11 Mar 2005 11:15:51 +0100
From: Edward Banfa [EMAIL PROTECTED]
Subject: [Asterisk-Users] NuFone Configuration [problem]
To: 'Asterisk Users Mailing List - Non-Commercial
Hello,
Could anyone say if there is any significant boost in voice quality with
the commercial SJPhone (payed for) vs. their free version?
Also, any reports on SIPPS free vs commercial?
It is worth to buy the licenced SJPhone/SIPPS to increase the voice
quality (they want $95 for it, pretty
Hi,
Does anyone know if it's possible to hook an asterisk PBX up to skype?
And if so, any config examples?
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To UNSUBSCRIBE or update
You mean that if on a certain queue, your agents are using SIP or IAX
phones, and you want to do a check so that when a cllers tryies to get into
the queue, if no agent is logged in, do something else with the caller
instead of hanging up?
Is that what you are trying to do?
-Original
Conference lock and member name been recorded and announced when they get in
and out of a conference is already available.
Check the wiki and look for meetme, you will see they have some parametes
like m,a,s that will help you control this features.
Anton Krall
Never mind.. answered my own question looks like their is a bounty
on the ability to do this :P
On Sat, 12 Mar 2005 18:28:00 -0500, Matt [EMAIL PROTECTED] wrote:
Hi,
Does anyone know if it's possible to hook an asterisk PBX up to skype?
And if so, any config examples?
Title: RE: [Asterisk-Users] Advanced conference features, meetme2?
Id like to know this to, Im prepared to kick in $50 to start off a bounty if no one else has done so.
Cheers,
Dean
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On
On Sat, 12 Mar 2005, Laurent Tostain wrote:
Hi,
We did an interconnection with our carrier few days ago. But, I noticed
that there was a signaling problem on our trnuk. In fact, Asterisk indicates
that the call is answered when we received ALTERTING message from our
carrier. This
I have grouped two capi controllers using * as the outgoing msn.
when the first two channels are busy normally and I try to use a third channel
the channels from the second controller are used.
But when a channel is occupied by the capisuite fax and we need a third channel
asterisk responds
Hi,
I've read the wiki... but would like some input from users here (not
implying that wiki writers aren't users).
I'm looking for a cheap (sub 60$) wired phone, or ATA device.. can
anyone recommend one (or several), and possibly a source?
___
as asked.
--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft.
You may not have most recent CVS. You should have this in your sip.conf:
rtcachefriends=yes
; Cache realtime friends by adding them to the internal list
; just like friends added from the config file only on a
; as-needed basis.
rtnoupdate=yes
; do not send the update request over realtime.
* is playing invalid message twice.
I have:
[internal]
include = outgoing ; include the outgoing context
include = voicemail
...
include = invalid
[invalid]
exten = _.,1,Answer
exten = _.,2,Playback(pbx-invalid)
exten = _.,3,Hangup()
asterisk is playing invalid message twice, WHY?
-- Executing
First, I did my noob homework and found a thread where this had been
discussed to some degree in a thread titled asterisk based bbs in
2004. I've got a question or two that that thread didn't address.
I want to set up a voice based BBS. This will barely stray from the text
based bulletin board
I am getting the following on dial-out via Sipphone to a 1-800 number
(numbers obscured):
-
== Spawn extension (macro-sipphone, s, 3) exited non-zero on
'SIP/eric-9546' in macro 'sipphone'
== Spawn extension (default, 1747xxx, 1) exited
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