> > Also I'd like to have worldwide users appear to be on the local
> >phone system through voice over ip. Is that possible? Our local
> >network is 100 mb/s and our internet connection is 768kb/s in both
> >directions. Would that be enough?
SIP users need to be on the same side of the f
Hi everyone,
Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a
problem and I think X-Lite is not even trying to contact SIP proxy while dialing.
Tomica
This mail was sent thru ZGWireless free network - www.zgwireless.net,
Internet connection sponsored by Isk
Skinny is not well supported by Asterisk. This was the answer when I asked
the same question few days ago.
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, October 18, 2003 4:42 AM
Subject: [Asterisk-Users] chan_skinny & XML Files for 7920
> Hi,
>
>
Tomica Crnek wrote:
Hi everyone,
Has anyone experience with xten.net's X-Lite SIP softphone and asterisk? I have a problem and I think X-Lite is not even trying to contact SIP proxy while dialing.
Tomica
This mail was sent thru ZGWireless free network - www.zgwireless.net,
Internet connect
Hi!
> Has anyone experience with xten.net's X-Lite SIP softphone and
> asterisk? I have a problem and I think X-Lite is not even trying to
> contact SIP proxy while dialing.
1. Read the FAQ on the xten site, the have good documentation over there.
Also read their PDF manual.
2. If X-Lite displ
Philipp von Klitzing wrote:
3. There is a bug in the currently available build 1079 (version 1) of
X-Lite where no matter which RTP ports you specify X-Lite will analyse
your NAT/firewall and select the ports by itself. In other words: X-Lite
doesn't care what range you enter as RTP ports. Pres
Hi all,
Is there any way to get * to start when linux boots?
I am running Red Hat 8.0, but a remote site I am testing IAX with has power
problems and the server there keeps re-booting, would be nice if everything
started up again automatically.
I noticed this in the list the other day,
I sugges
David J Carter wrote:
Hi all,
Is there any way to get * to start when linux boots?
I am running Red Hat 8.0, but a remote site I am testing IAX with has power
problems and the server there keeps re-booting, would be nice if everything
started up again automatically.
I noticed this in the list the
Cheers,
Do I add the safe_asterisk to the rc.local file?
You may tell I am new to Linux.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: 18 October 2003 10:40
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Auto Start
David J Carter
Hi,
I am trying to write an AGI script that executes a
shell command in C ie. ls. I tried VERBOSE AGI command
or to send the "!command ls" to stderr but the command
does not execute it just displays on asterisk console.
Does anybody has any idea about it?
Thanks
Dimitris
__
David J Carter wrote:
Cheers,
Do I add the safe_asterisk to the rc.local file?
You may tell I am new to Linux.
Dave
Yes, rc.local should do it for you..
Don't worry about being new to Linux, I have been using it for a few
years and I am still learning, I still think I am a newbie.. :)
__
I did this, but ok, I'll try again
- Original Message -
From: "WipeOut" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, October 18, 2003 11:01 AM
Subject: Re: [Asterisk-Users] x-lite
> Tomica Crnek wrote:
>
> >Hi everyone,
> >
> >Has anyone experience with xten.net's X-Lite
I have put ./var/sbin/safe_asterisk in the rc.local file but it still
doesn't start.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: 18 October 2003 11:31
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Auto Start
David J Carter wrote:
>Ch
one more point,
infact we tried with 'callprogress=yes', then both of the extensions starts ringing,
but the callee can not hear the ringback...
any suggestions...???
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, October 17, 2003 12:45 PM
Subject:
David J Carter wrote:
I have put ./var/sbin/safe_asterisk in the rc.local file but it still
doesn't start.
Have you got the zaptel drivers loading at startup?
This can either be done by using modprobe commands in the rc.local or by
using the init script that comes with the zaptel source.. jus
Yup,
I did modprobe when I loaded everything. If I issue the reboot command then
I see the zaptel being unloaded.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: 18 October 2003 12:11
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Aut
On Wed, 15 Oct 2003 13:49:23 -0500 (CDT), Dave Weis wrote
> On Wed, 15 Oct 2003, M.A. Ali wrote:
> > I am kind of new to asterisk. Here is a little prolem that I am
facing.
> > Here is my problem and questions: I am just adding two gnophone
users to
> > my dialplan, all three systems are within lan
> I did this, but ok, I'll try again
If your box is behind NAT you need to tell xlite NOT to detect it -- I had
this problem where the * box and the xlite box were behind NAT and NAT was
not needed for xlite to talk to *, but xlite decided that it was so was
sending the wrong IP.
Everything el
I thought I'd post my experiences for the benefit of anyone else who may be
at the point I was when I first started with asterisk.
I have 2 incoming analog lines (north eastern U.S., Verizon) where one is
set to ring if the first is busy.
I bought a bare-bones system from abs-pc with the following
Title: Message
We have added an
Asterisk Forums to our existing web site. It will make things easier to search
for related problems, etc.
http://www.pbxtech.info/forumdisplay.php?f=113
On Sat, 2003-10-18 at 05:23, Jim Paraschou wrote:
> Hi,
>
> I am trying to write an AGI script that executes a
> shell command in C ie. ls. I tried VERBOSE AGI command
> or to send the "!command ls" to stderr but the command
> does not execute it just displays on asterisk console.
> Does anybody
I don't know if this is a problem with my cisco sip IP Phones or
asterisk but I thought I would post here in case someone else has
experienced this issue.
When I make a call from my SIP cisco IP Phone to some remote IVRs I
never get the rest of my soft keys, only the "End Call" soft key, and
also
Hi,
We would be interested in this project also.
Paulo Mannheimer wrote:
Hi All,
We've been developing for a while an IDE for Asterisk, and the time has
come to open it for beta testers.
You can check at www.instant.com.br/viv.html for a snapshot of the
application.
Current modules are Dial
Ok after find this post
http://www.mail-archive.com/[EMAIL PROTECTED]/msg07004.htm
l. I have figured out my issue and I have refined my question.
Does anyone how to make asterisk tell the calling sip-ua that the remote
party "answered" the phone as soon as it sees rtp coming in even though
the re
ok, ive just updated a server and now im getting these messages a lot on
the console:
-- Called g1/X
WARNING[42004]: File app_dial.c, Line 317 (wait_for_answer): Unable to
forward voice
WARNING[42004]: File app_dial.c, Line 317 (wait_for_answer): Unable to
forward voice
WARNING[
I bought a Grandstream 101, then I bought 2 more. I also got a Cisco
ATA186. I had looked into using the ATA186 with asterisk, and it looked
like I could get it to work. When I got it, I realized that It didn't have
the same firmware as I thought it would. In fact, as it was, I couldn't get
I crimped one using cat 5 cable since this is just a bench-test unit.
Thanks for the link, though.
Turns out this question is answered in the FAQ, but it's certainly much
easier to crimp one using Jared's pictures than the terse pin-out in the
answer.
asterisk # ztcfg -vvv
Zaptel Configuratio
On Sat, 2003-10-18 at 15:33, duncan wrote:
> ok, ive just updated a server and now im getting these messages a lot on
> the console:
Use ´cvs update -D date´ to rollback your source to the date you want
then check again to see if its new cvs problem.
--
Juanjo sin .sig
On Sat, 2003-10-18 at 15:17, David Harris wrote:
> Does anyone how to make asterisk tell the calling sip-ua that the remote
> party "answered" the phone as soon as it sees rtp coming in even though
> the remote IVR didn't "connect" the call.
>
> I found a couple of remote IVRs that when I call th
Hello,
I new to this, but with the help of mailing lists archives and IRC I am able
to build my PBX. Thanks to all who had help me to reach till here.
I am stuck at a point where I can't find the solution on mailing lists or
even on IRC.
I have individual 4 DID (Direct Inward Line) coming from
Juan,
That is a helpful discovery. If I too route the calls out of an IAX
provider (voicepulse) rather then my usual Cisco SIP Gateway I can get
DTMF through with the Cisco phone still not presenting the usual "soft"
key options.
So that means that the Cisco IP Phone IS sending DTMF to calls t
Title: Message
hi , i'm testing
asterisk like and Automatic attendant with a callmanager and vg200 gateway with
1 t1
everithing works
finw but some times asterisk didnt not disconnect calls and star growing the
number of
connections from
asterisk to callmanager , and when this connections
Howdy,
Does anyone know if there are any problems running Asterisk when using
later 7960 SIP versions like 04.04 or 05.03?
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED]
phone: 650-207-9855
fax: 877-408-0105
___
Asterisk-Users mailing list
[EMAIL PR
I have googled this one to death, and can't find anything.
I added a number of new users to my asterisk (current CVS) system. I am
using the "Voicemail2" family.
I added entries in extensions.conf and voicemail.conf for my new users,
and I have tested leaving and retrieving new voicemails for
I have been trying to get the DTA310 to work properly with Asterisk for the
last week. It seems to connect but it does not play back any sound and I
cannot dial it by using x-lite. Sip debug looks pretty good. I was
wondering if someone has a working config that they could post so that i
could s
> > How does the CISCO ATA sound quality, functionality and stability
compares
> to the Grandstream phones?
Sound Quality using G.729: Grandstream phones are superior. They sound
perfect even with slight packet loss. The ATA will sound very good with 0%
packet loss but if you ramp it up a bit i
On Sat, 2003-10-18 at 17:25, Paul Mahler wrote:
> Howdy,
>
> Does anyone know if there are any problems running Asterisk when using
> later 7960 SIP versions like 04.04 or 05.03?
I have 4.4 running without problems.
--
Juanjo sin .sig
___
Ast
** REPOST: A week later and no feedback - am I the only one
** who'd find this functionality useful? No other AGI stuff
** out there needing something similar?
I'd like some feedback on potentially submitting a request (and probably a
patch too) to change the way the AGI command GET DATA works.
R
This is the same problem that I've posted previously about "early
audio". Depending on the device and your particulary NAT setup, early
does and doesn't work. IAX does pass the DTMF with Cisco ATAs, but with
other devices it does not. Still haven't managed to solve this one
except with DTMF set
Brian Capouch wrote:
I have googled this one to death, and can't find anything.
I added a number of new users to my asterisk (current CVS) system. I
am using the "Voicemail2" family.
I added entries in extensions.conf and voicemail.conf for my new
users, and I have tested leaving and retrievi
I made a post a few days ago with a patch to the queue to allow you to insert
a penalty value in dynamic queue adding.
http://asterisk.650dialup.com
Since then, I decided I would make a whole new strategy called 'skillbased'
I added a new option skillmask to the Queue appl and the
AddQueu
Hi,
Just back from the Rugby match, (we won), and am now trying again to get
asterisk to auto start without being there when the server reboots.
Wipe Out asked:
Have you got the zaptel drivers loading at startup?
This can either be done by using modprobe commands in the rc.local or by
using the i
Dave,
After the system boots up, you check to see which modules are loaded by
doing a "lsmod". zaptel and others that you need should be listed. If not
you can manually add "modprobe zaptel" and the other drivers into your
rc.local file.
Paul
> Just before the Logon prompt appears on boot I see
WipeOut wrote:
If you are using VM2 and using something other than the [default]
context then you will not create the correct directory structures when
you run the addmailbox script to create the mailboxes..
I have attached a copy of my addmailbox2 script which takes into account
the context w
** REPOST: A week later and no feedback - am I the only one
** who'd find this functionality useful? No other AGI stuff
** out there needing something similar?
I'd like some feedback on potentially submitting a request (and probably a
patch too) to change the way the AGI command GET DATA works.
Rig
While that change is fine, you could also just write the same
functionality with get digit and deal with it inside the AGI app.
On Sat, 2003-10-18 at 16:50, Paul Crick wrote:
> ** REPOST: A week later and no feedback - am I the only one
> ** who'd find this functionality useful? No other AGI stuff
> While that change is fine, you could also just write the
> same functionality with get digit and deal with it inside
> the AGI app.
I guess the digit handling part is doable with WAIT FOR DIGIT.. I was going
to ask about interupting the prompt that played before that but on RTFM-ing
I see that S
More appropriately, Skinny is still considered beta quality in Asterisk.
Patches are welcome and encouraged.
Jeremy McNamara
Tomica Crnek wrote:
Skinny is not well supported by Asterisk. This was the answer when I asked
the same question few days ago.
- Original Message -
From: <[EMA
> I now have only ./safe_asterisk in my rc.local file.
>
Another way to start/monitor * is to put this in /etc/inittab:
# Run asterisk in runlevels 2-5
A1:2345:respawn:/usr/sbin/safe_asterisk
If * dies for any reason, its automatically restarted within about
five minutes. (It also cause
> While that change is fine, you could also just write the
same functionality with get digit and deal with it inside
the AGI app.
I guess the digit handling part is doable with WAIT FOR DIGIT.. I was going
to ask about interupting the prompt that played before that but on RTFM-ing
I see that STR
I've reading this lists few months.
We are small company, that makes some system intregration, development and deployments
in VoIP scene.
Completely under linux. Today i have 6 machines with asterisk, huge test base -
including devices like AS5350, Audiocodes gateways,
ATAs, IP phones ...
Now is
> How stable are these channel drivers?
I haven't run into any real stability issues, then again I just have
4 grandstreams sitting on a desk. :)
Oh, and a CAC AB1 channel bank connected to a Digium T100P.
> Is there any commercial support for faster bugcleaning, fixing ...
> (anything will be i
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