Dear all
I have video confranceing deivice Codian and i want to
intergrate asterisk box with codian so voice confrance is possible with codian
users means some users have not codian endpoint so thay call join confranceing
with SIP PHONE
I have configures asterisk and
dear all
I there any feature of huntgroup in asterisk means when i call
on huntgroup number then any available phone in that group rining is there any
feature like this ???
Rgd
satish patel
-
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Hi,
I am going to be on the road for the next few days and with the
variable delay on the mailing list, I am posting this now, 4 days
before the conference. If you haven't yet listened or participated,
please consider doing it. We have a great kernel of people at all
levels of expertise and ideas
Hi,
I am able to dial through asterisk PBX having TE120P card to E1 card
running application. Communication was established successfully
Now, I want to do the reverse way out. I am using the following
configurations
1)zaptel.conf
span=1,1,0,ccs,hdb3,crc4
defaultzone=us
Sanchal,
You may want to make sure that you have immediate=no set for your E1 channels
in zapata.conf. This makes asterisk wait for digits, rather than skipping to
the s extension on incoming calls.
--TS
[EMAIL PROTECTED] 7/30/2007 4:14 AM
Hi,
I am able to dial through asterisk PBX
I am running asterisk 1.2 with bristuff 0.3.0 and have the following
problem:
When I make a call out it fails with a chanunavail message but if I make a
call in and then make a call out it is successful. I think this is because
BT set the Layer 1 to turn off after a period of time.
I need to
Hi:
I want to have conference call with asterisknow and need 2 ports E1.Which
Digium card is better?TE212 or TE220.I haven't problem with motherboard.
Regards.
-
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Hi,
I am able to dial through asterisk PBX having TE120P card to E1 card
running application. Communication was established successfully
Now, I want to do the reverse way out. I am using the following
configurations
1)zaptel.conf
span=1,1,0,ccs,hdb3,crc4
defaultzone=us
On Mon, Jul 30, 2007 at 10:09:32AM +0100, asterisk wrote:
I am running asterisk 1.2 with bristuff 0.3.0 and have the following
problem:
Which version of bristuff do you have exactly?
asterisk -rx 'zap show version'
When I make a call out it fails with a chanunavail message but if I make a
Hi,
I'm brazilian. By the way, Why such a question?
See you.
Ronaldo.
Jozeph Brasil wrote:
Some brazilian here on list?
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Hi list
I've written a tool that works as a lightweight (standalone - no asterisk)
balancer for IAX servers. It's in early development now, but seems to be stable
enough and handles couple hundred simultaneous calls with not much latency
(SIPp + asterisks tested).
It's configurable by listing
Hi all,
I have a Wildcard TE110P connected to a E1 line an I want to reserve
channels in the following way:
channels 1-15 and 17-21 for incoming calls
channels 22-28 for outgoing calls
channels 29-31 for emergency calls
My zaptel.conf looks like this:
; incoming
group = 1
signalling=pri_cpe
Hi,
I would like to know if one can set the outgoing
caller ID within Asterisk when calls are going out
through:
1) an analog POTS line (I suppose not)
2) a telco BRI line (I don't think so)
3) a telco PRI line (maybe)
4) a voip provider (surely)
Thanks,
Vieri
Hi guys,
I've setup on box with a TE110P and time to time I need to access remote
equipment outside of our office and use a data channel. I'm wondering if do
I need to buy a POTS line only for this time to time acess or what's the
easiest way to do that via my TE110P on asterisk box.
I know that
Yep! From São Paulo - SP
Where we can help?
Regards
Josué
2007/7/30, Ronaldo [EMAIL PROTECTED]:
Hi,
I'm brazilian. By the way, Why such a question?
See you.
Ronaldo.
Jozeph Brasil wrote:
Some brazilian here on list?
___
0.3.0-pre-1s
After working with traditional pabx's in the past I have known the setting
of layer 1 to call has fixed this problem.
Thanks
Neil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: 30 July 2007 11:34
To:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
satish patel wrote:
dear all
I there any feature of huntgroup in asterisk means when i call on
huntgroup number then any available phone in that group rining is
there any feature like this ???
You can use queues for this purpose.
Barry
- Tzafrir Cohen [EMAIL PROTECTED] wrote:
Interesting. One thing thoough: what's the license of your code?
It's MIT - I forgot to add that. I'll stick the banners to files soon, with
next update to the package. (along with some fixes, etc)
Stanisław Pitucha
Gradwell Dot Com
I got my G729 licenses installed.I can make calls out and receive
calls and the system shows the licenses are in use, however, if I try
to call voicemail.. the CLI shows the files are playing, however I
don't hear anything.
___
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You sure about that?
Having a config that looks like this:
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=g726
context = from-sip-external ; Send unknown SIP callers to this context
callerid
On 7/30/07, voiplist [EMAIL PROTECTED] wrote:
I noticed that if I have an agent logged in using AgentCallBackLogin
and that agent is unreachable for some reason (SIP phone unplugged)
calls to him/her will completely yack.
For example:
1-Agent 500 is the only one logged into queue number 1.
On Mon, 2007-07-30 at 02:40 -0700, fateme fatah wrote:
I want to have conference call with asterisknow and need 2 ports
E1.Which Digium card is better?TE212 or TE220.I haven't problem with
motherboard.
There are two major differences between the TE212P and the TE220 cards.
The first is the
Hi All,
I would like to build a little announcement server with asterisk.
Is it possible to do the following:
- when * gets the INVITE message, it should send 183 Session in progress
back
- it should play an announcement message in early media
- then, redirect the client to a specified URI
Hi,
Reading from various Netfilter mailing lists, I'm wondering whether or not,
has anyone ever got a successful experience with SIP conntrack and Asterisk.
For instance, this feature was :
- introduced in Linux kernel 2.6.16,
- improved in 2.6.18
- enhanced in 2.6.22
- I even read something
Hello all,
Where can I find a list of description for each sound files provided by the
asterisk-sounds-main Debian package? You can find the contents of my
/usr/share/asterisk/sounds/ directory at http://paste.debian.net/33679.
Thank you in advance.
GNUbie
On Sun, 2007-07-29 at 14:51 +0100, Thomas Kenyon wrote:
iptables -A PREROUTING -t nat -p tcp -i eth0 --dport 4569 -j DNAT --to
ip-of-asterisk-box:4569
should work, assuming you have the relevant parts compiled in.
Just for your information, IAX traffic is UDP, not TCP. I just thought
I'd
Hello,
I'm looking for a way to play sound file, and control the playback
trough web interface. Is it possible to use AGI to play a sound file
and then by receiving some event stop playing it, and play another
file. The catch is that i want to seek to 1st minute, 5th minute, etc
- so regular
Hello,
I have been reading up on the capabilities of the Asterisk-Java library. I
believe that this library can act as an interface between a Java GUI(custom
softphone) and the Asterisk server. Seems like the Live API would be easiest
to use to make the connection to the Asterisk server and
Isso nao vai parar?
On 7/30/07, Josué Conti [EMAIL PROTECTED] wrote:
Yep! From São Paulo - SP
Where we can help?
Regards
Josué
2007/7/30, Ronaldo [EMAIL PROTECTED]:
Hi,
I'm brazilian. By the way, Why such a question?
See you.
Ronaldo.
Jozeph Brasil wrote:
Some
On Mon, 2007-07-30 at 21:45 +0800, GNUbie wrote:
Where can I find a list of description for each sound files provided
by the asterisk-sounds-main Debian package?
The file core-sounds-en.txt should contain the text of each of the sound files.
--
Jared Smith
Community Relations Manager
Digium,
Chineese now in asterisk mailing list?
Ary Junior a crit:
Isso nao vai parar?
On 7/30/07, Josu Conti [EMAIL PROTECTED] wrote:
Yep! From So Paulo - SP
Where we can help?
Regards
Josu
2007/7/30, Ronaldo [EMAIL PROTECTED]:
Hi,
I'm brazilian. By the way, Why such
As a different approach, QueueMetrics includes a perl script that does the
real-time uploading of queue_log data into a database. It is being used in
a large number of high load installations worldwide, so I'd say it's a
pretty proven solutions, and it's very lightweight. As an added bonus,
On Mon, Jul 30, 2007 at 01:48:12PM +0100, asterisk wrote:
0.3.0-pre-1s
After working with traditional pabx's in the past I have known the setting
of layer 1 to call has fixed this problem.
There have been quite a few updates to briistuff since.
and if all of this doesn't help, maybe try
can you explain me how it will work caz i have not much idea about asterisk i
am beginner so can u explain me how to use queue and how to forward my call to
huntgroup
Barry L. Kline [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
satish patel wrote:
dear all
I there
JMF ( http://java.sun.com/products/java-media/jmf/ ) for audio... a good
example to use JAIN SIP and JMF is the SIP Communicator source code (
https://sip-communicator.dev.java.net/ ) ...
[]s
Ary Junior
On 7/30/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello,
I have been reading up on
On Mon, Jul 30, 2007 at 09:45:10PM +0800, GNUbie wrote:
Hello all,
Where can I find a list of description for each sound files provided by the
asterisk-sounds-main Debian package? You can find the contents of my
/usr/share/asterisk/sounds/ directory at http://paste.debian.net/33679.
It's
GNUbie wrote:
Hello all,
Where can I find a list of description for each sound files provided by the
asterisk-sounds-main Debian package? You can find the contents of my
/usr/share/asterisk/sounds/ directory at http://paste.debian.net/33679.
You would have to contact the person that built
Yes. Use the group= setting in zapata.conf. group=1 then
Dial(Zap/g1/5551212)
satish patel wrote:
dear all
I there any feature of huntgroup in asterisk means when i call
on huntgroup number then any available phone in that group rining is there
any feature like this ???
If your provides has not provisioned any channels on your t1 as data
then this wont work. im guessing that for wha you want an FXS post
will do
On 7/30/07, Marco Mouta [EMAIL PROTECTED] wrote:
Hi guys,
I've setup on box with a TE110P and time to time I need to access remote
equipment outside
On Mon, 2007-07-30 at 07:01 -0500,
[EMAIL PROTECTED] wrote:
Date: Mon, 30 Jul 2007 12:19:13 +0100 (BST)
From: Stanis?aw Pitucha [EMAIL PROTECTED]
Subject: [asterisk-users] Lightweight IAX balancer
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type:
Jack wrote:
Hi all,
I have a Wildcard TE110P connected to a E1 line an I want to reserve
channels in the following way:
channels 1-15 and 17-21 for incoming calls
channels 22-28 for outgoing calls
channels 29-31 for emergency calls
My zaptel.conf looks like this:
; incoming
group
1 No 2 I dont know. 3 Currently in the us the answer is yes
On 7/30/07, Vieri [EMAIL PROTECTED] wrote:
Hi,
I would like to know if one can set the outgoing
caller ID within Asterisk when calls are going out
through:
1) an analog POTS line (I suppose not)
2) a telco BRI line (I don't
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ahoy
I'm trying to setup Asterisk on debian etch (with the debian packages)
with a Fritz!Card PCI ISDN card and chan_capi.
Everything seems to be configured the right way (excerpts below),
Asterisk seems to see the ISDN-card but if i try to place a
hi all,
I am trunking via iax2 2 asterisk serverses
if both of them have static ip addresses, I can connect them using no
password, password or auth rsa with a pair of keys.
If one of them has dynamic ip address and need to register on the other
server, I can connect them with no password, but I
On Mon, Jul 30, 2007 at 02:01:49PM +0200, Jack wrote:
Hi all,
I have a Wildcard TE110P connected to a E1 line an I want to reserve
channels in the following way:
channels 1-15 and 17-21 for incoming calls
channels 22-28 for outgoing calls
channels 29-31 for emergency calls
My
On Mon, 2007-07-30 at 16:46 +0200, Florian Arthofer wrote:
Shouldn't i see _something_ on the console, even if
the DID which is dialed isn't configured yet?
Unfortunately, I don't think so. You might want to add a pattern match
to your dialplan that would match any DID, and see if that helps.
On Mon, Jul 30, 2007 at 10:40:57AM -0400, C F wrote:
On 7/30/07, Vieri [EMAIL PROTECTED] wrote:
I would like to know if one can set the outgoing
caller ID within Asterisk when calls are going out
through:
1) an analog POTS line (I suppose not)
2) a telco BRI line (I don't think so)
Hi All,
In our small office calls to the PSTN are currently sent via Asterisk and a
Linksys SPA3102 (1 x FXO and 1 x FXS):
SIP Phone -- Asterisk -- Linksys SPA3102 -- PSTN
If the PSTN is in use on SPA3102 I need a way to get the call to then route
out over IAX termination.
Having some issues with getting sound from a call.
I have 4 systems. 3 main systems which handle calls for our 3 locations.
The 4th system is the central voice mail system. When an inbound call
gets passed to someones voice mail its done with an IAX2 connection. The
same happens after hours
I got my G729 licenses installed.I can make calls out and receive
Make sure you add g729 to the voicemail config as well.
John
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Make sure you add g729 to the voicemail config as well.
?? Don't understand. I still want my format=wav|gsm.But that
doesn't seem to be the issue... as I can't even hear the password
prompts.
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On Mon, 30 Jul 2007 05:24:31 -0700 (PDT), Vieri wrote:
Hi,
I would like to know if one can set the outgoing
caller ID within Asterisk when calls are going out through:
1) an analog POTS line (I suppose not)
Nope
2) a telco BRI line (I don't think so)
3) a telco PRI line (maybe)
4) a
I've got the ztdummy kernel module loaded and seem to have all the desired
prerequisites in place, but Asterisk never seems to compile with MeetMe()
application support enabled, nor does there appear to be a module I am
failing to load that would contain this application.
Is there something
On 7/30/07, Jared Smith wrote:
Just for your information, IAX traffic is UDP, not TCP. I just thought
I'd bring that up so that someone didn't mistakenly open up their
firewall for TCP traffic instead of UDP traffic and wonder why IAX
traffic wasn't making it through.
Amen !
I had
Hi,
We are running asterisk 1.2.16 and need to connect two channels which
are already established. We are currently using app_meetme to achieve
that, but we are sometimes unhappy, as app_meetme provides functionality
that produces load that we do not need in our two party conferences. I
El Sun, Jul 29 de 2007 a las 20:04 +0800, Steve Underwood comentaba:
What versions of software did you use to get a screwed up result like
that? The message Don't know how to handle signalling event Accepted
is printed at the end of a case statement which does handle that event.
I the
Hi,
what does your modules directory contain? Can you find a file
/usr/lib/asterisk/modules/app_meetme.so after make install?
Knud
Alex Balashov schrieb:
I've got the ztdummy kernel module loaded and seem to have all the desired
prerequisites in place, but Asterisk never seems to compile
Hello All,
I am almost done with my notifications system, but I am stuck with
prompting the correct time.
I went over the phpagi doc's, on how to say a given time using SAY
TIME time escape digits.
According to http://www.voip-info.org/wiki/view/say+time it say time
is number of seconds
On Mon, 30 Jul 2007, Knud Müller wrote:
what does your modules directory contain? Can you find a file
/usr/lib/asterisk/modules/app_meetme.so after make install?
No. I know it needs to be compiled, but it is not being compiled no
matter what I seem to do in the way of arguments to
I can't even hear the password prompts.
Ahh... have you loaded the G729 sounds? Are you getting errors in the logs?
John
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I don't have any g729 sounds loaded.. they are just the gsm sounds...
shouldn't asterisk do the conversion.. although at a license hit?
On 7/30/07, John Faubion [EMAIL PROTECTED] wrote:
I can't even hear the password prompts.
Ahh... have you loaded the G729 sounds? Are you getting errors in
At 21:59 7/29/2007, Paul Hales wrote:
I even got a Polycom here saying I'll be back which was funny for
about an hour, then not funny at all.
PaulH
Kewwl! How do you get the .wav files into the Polycom?
On Fri, 2007-07-27 at 12:36 +0800, James Andrewartha wrote:
Hi all,
Has
I have a support call AGI script that has been working
flawlessly for a couple of years now. It dumps the customer into a
MeetMe conference room, then dials a bunch of support engineers,
and connects anyone who accepts the call into the conference room.
The conference room is recorded. After
Hello,
in your sip.conf do you have
[yourprovider]
username=
fromuser=
secret=
host=another.server.com
nat=yes
.
.
.
.
and in your extensions.conf
And the extensions.conf:
...
exten = _X.,1,Dial,SIP/yourprovider
...
Best Regards
sip:[EMAIL PROTECTED] )
On 7/29/07, Ary Junior [EMAIL
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
satish patel wrote:
can you explain me how it will work caz i have not much idea about
asterisk i am beginner so can u explain me how to use queue and how
to forward my call to huntgroup
http://www.orderlyq.com/asteriskqueues.html
Barry
Hi Steve,
The following packets match the offending warnings on the * console by
time and number of occurrences and . The SRC and DST ports vary between
calls (naturally) but the rest remains the same.
Frame 3 (60 bytes on wire, 60 bytes captured)
Ethernet II, Src: Grandstr_0b:a5:3e
Hello,
I got a SPA3102 and everything works fine except calling from voip to phone
on fxo port. The phone ring but doesn't get any sound. I connected SPA at my
asterisk server and i want to call from asterisk through SPA to fxo port
where i have a regular phone. Thank you for support.
I've been investigating an issue on the Asterisk bugtracker recently:
http://bugs.digium.com/view.php?id=10300
The reporter shows that there are places in the code where strings are
truncated. You can read the bug report for full information. I suspect
that the problem is specific to the
Why would you want to do that? let Asterisk (using zap/g in app_dial)
take care of which channel are used for outbound but assign all the
channels to that g, reject any incoming calls if there are already 7
incoming active calls with a congestion PRI_CAUSE. Do the same for 20
outgoing active
Hello,
I am trying to transmit and receive sound over IP using the Java Media
Framework(JMF) RTP. I was wondering if its possible to create an RTP Stream
from my own computer and assign it to a URL. If anyone knows how I would do
this, could they point me to some instructions or an example.
Am Montag, den 30.07.2007, 05:24 -0700 schrieb Vieri:
Hi,
I would like to know if one can set the outgoing
caller ID within Asterisk when calls are going out
through:
1) an analog POTS line (I suppose not)
2) a telco BRI line (I don't think so)
3) a telco PRI line (maybe)
4) a voip
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
Lee.
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On Mon, 2007-07-30 at 14:29 -0700, Lee Howard wrote:
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
This was added as a April Fools joke, and has since been removed. It's
nice to know that even software engineers have a sense of humor from
time to time. :-)
Greetings all,
When using QueueAdd via the dialplan app, we are able to define an agent
name... however, I don't see how this can be done via the Asterisk
Manager. Am I missing something, or is this just not possible?
Regards,
Jeff
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Quoting Lee Howard [EMAIL PROTECTED]:
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
I have to agree, there comes a time when someone has to say no to
stuff that has no business being in production software.
Remember when an o/s fit on a floppy with room to
Lee Howard wrote:
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
Lee.
It was done as a joke. It was committed only to trunk, and was only
compiled if explicitly enabled.
Mark seemed to get a kick out of it, so, yes, I guess you could say it
was useful
Relax, its only in trunk.
Zoa
Lee Howard wrote:
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
Lee.
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To
Jon Pounder wrote:
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
I have to agree, there comes a time when someone has to say no to
stuff that has no business being in production software.
Well, you'll have to excuse him for trying to make a joke. :)
On Mon, Jul 30, 2007 at 02:29:32PM -0700, Lee Howard wrote:
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
Ghod... nobody has a sense of humour anymore. :-)
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
On Mon, 30 Jul 2007, Jay R. Ashworth wrote:
On Mon, Jul 30, 2007 at 02:29:32PM -0700, Lee Howard wrote:
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
Ghod... nobody has a sense of humour anymore. :-)
Might just be making a slippery slope argument.
Jay R. Ashworth wrote:
Ghod... nobody has a sense of humour anymore. :-)
I know. I better not list all of the other things we have done as a joke.
Someone might have heart failure. ;)
--
Russell Bryant
Software Engineer
Digium, Inc.
___
try adding the line
MemberName : name
On 7/30/07, Jeff Iddings [EMAIL PROTECTED] wrote:
Greetings all,
When using QueueAdd via the dialplan app, we are able to define an agent
name... however, I don't see how this can be done via the Asterisk
Manager. Am I missing something, or is this
On Mon, Jul 30, 2007 at 05:42:52PM -0400, Jon Pounder wrote:
Quoting Lee Howard [EMAIL PROTECTED]:
http://www.asterisk.org/node/48327
I mean, really... you're kidding me, right?
I have to agree, there comes a time when someone has to say no to
stuff that has no business being in
After:
Action: QueueAdd
I presume? Thanks!
0xception wrote:
try adding the line
MemberName : name
On 7/30/07, *Jeff Iddings* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Greetings all,
When using QueueAdd via the dialplan app, we are able to define an
agent
That did the trick, thanks!
Question, where did you find that documented? :)
Jeff
0xception wrote:
try adding the line
MemberName : name
On 7/30/07, *Jeff Iddings* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Greetings all,
When using QueueAdd via the dialplan
Has anyone set up Speechphone (Mandi) directly with Asterisk and not used an
ATA? If so, could you share how you did it?
TIA
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Very low chances for that module if any. I haven't been using OpenSER much and
I don't think I'll be using it soon - but who knows. Let's hope that
implementation will be clean enough to turn it into a library easily if someone
else wants to do it one day.
So far another pack is available at
you know... i have no clue...
i have it in my code somewhere. so i must of found it someplace. possibly in
the phpagi-manager code. maybe some other random place. most asterisk info
is scatter about, mixed up, and often out of date. so it's really really
hard to tell some times.
On 7/30/07, Jeff
Aye. I'm not one to ask without doing a bit of research and I couldn't
find that anywhere... even tried to figure it out by looking at the
code. You're the best. Thanks again.
Jeff
0xception wrote:
you know... i have no clue...
i have it in my code somewhere. so i must of found it someplace.
On Mon, 2007-07-30 at 16:09 +0100, Chris Blunt wrote:
If the PSTN is in use on SPA3102 I need a way to get the call to then
route out over IAX termination.
Usually, the best way to accomplish this is to send a call to your
Linksys ATA by using the Dial application from the dialplan, and then
1. No
2. No
3. Only if your particular provider's switch allows it. Most will allow
numbers to be set, but block the call if you try to set name.
4. Yes.
Yours,
Michael Munger, dCAP
404-438-2128
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Just so people on the list can search later: I found the solution:
The smoothwall we have as our firewall / router needed to be reset. It
went haywire and wasn't forwarding anything after about the 5th entry. I
deleted everything out of the web interface for port forwarding,
confirmed it went bye
Following up on my own post, and not quoting myself (tsk, tsk), I
found a forum thread on Google that discussed a similar problem.
They claimed it was a SIGHUP being sent to the script when the
caller hung up, even though DeadAGI shouldn't get that type of signal.
Anyway, it turns out that was
dear all,
when i complied the latest Zaptel-1.2.19 to upgrade my asterisk system,
it told me those errors:
cc -c -fPIC -DSTANDALONE_ZAPATA -DBUILDING_TONEZONE -DHOTPLUG_FIRMWARE
-I. -O4 -g -Wall -DBUILDING_TONEZONE -o zonedata.lo zonedata.c
In file included from zaptel.h:31,
from tonezone.h:27,
I am using TE212P with asterisk-1.2.18. It has echo DTMF in hardware to
support. I use it on Dell Power Edge 85 no IRQ's ...
Ya, just make sure that u get a good card I got the a broken card first time
which ddnt work for echo cancellor then RMA'ed it with new one.
--
Deepak
Hello Tzafrir,
On 7/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
It's in /usr/share/doc/asterisk-sounds-main/sounds.txt.gz , as you
should have expected (documentation for package foo normally resides at
/usr/share/doc/foo/ and text files that are long enough are gzipped).
I already
Hi:
I want to have conference call service.You offer me use asterisk or
asterisknow.
Regards.
-
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Hi,
Is there any Royalty one needs to pay when using the inbuilt exisimg
asterisk on hold music or when using any other mp3 from a music album.
I think we need to pay for the later, but I am not sure if we need to pay for
the inbuilt asterisk(freepbx) on hold music.
--
You can use both Asterisk or AsteriskNow to have meetme (conference room)
On 7/30/07, fateme fatah [EMAIL PROTECTED] wrote:
Hi:
I want to have conference call service.You offer me use asterisk or
asterisknow.
Regards.
--
Be a better Globetrotter. Get better
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