e the MB and do a full re-install
(again).
Gratefully,
JonathanOn 9/13/06, Jonathan Barratt <[EMAIL PROTECTED]> wrote:
Thanks for the reply Steve.
I am calm now. :)
I've been getting the exact time and number of the dropped calls for
the last couple weeks, and there was nothing in sy
e them all myself with research and
experimentation, except for this persistent intermittent dropped call
problem...
I'm really grateful for your input Steve, please keep it coming!
Thanks very much!
Jonathan
On 9/13/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
Sorry, see now that it is
#x27;s no heavy network traffic going on besides Asterisk to the phones (Aastra 480i's).
What other factors can I investigate?
This client is so unhappy they are ready to go back to their old PBX system.
I am desperate, please help!!
Thanks
!\n", 9) = 9
So the problem is the last read call, which differs from all the
previous read calls fxotune makes in that the second parameter is a hex
value rather than a quoted string.
Unfortunately I am clueless as to where to go from here.
Any further suggestions would be grea
the same network and setup we never experience delays.Any suggestions/help would be much appreciated.
Thanks,Jonathan-- Jonathan Palley | Idapted Inc.[EMAIL PROTECTED]
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ed to clear up the echo in this office.
Thanks in advance,
Jonathan
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Title: RE: [asterisk-users] Polycom 501 config questions
Dumb question here: Why the
need to dial 9 for an outside line? If your extensions are less than 7 digits
long then you know anything "XXX." is an outside call
Maybe this isn't true everywhere, just
curiou
I’ve only used a Quintum a few
times,sorry.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul
Sent: Friday, August 25, 2006 6:49
AM
To:
Asterisk-Users@lists.digium.com
Subject: RE: [asterisk-users]
quintum Calling Card
Hello Jonathan,
I tried in
Abdul, it doesn’t sound like you
need to do anything to the Quintum. I would recommend making your dial plan execute
the AGI script of your choice no matter what number is dialed from the context
where the quantum users land.
-Jonathan
From: [EMAIL PROTECTED]
[mailto
chan_sip.c:
update_call_counter(101) - decrement call limit counter
Anyone have any ideas on this?
-Jonathan
Jonathan Creasy
Network Engineer
BluegrassNet Development
www.bgnd.com www.bluegrass.net
o. 502-589-4638
c. 502-889-5567
h. 81
I was wondering which of these cards would be better for a 1-2
line SOHO. I would like room to grow as well
as I am concerned with voice quality and life expectancy of the product. Any
input into which one and why would be greatly appreciated.
Thanks,
Jon
_
Title: Re: [asterisk-users] Quick One - PHP Script to restart Asterisk
execute "asterisk -rx
'restart when convienent";
?>
Not the exact syntax but should be enough
to get you going.
From: [EMAIL PROTECTED] on
behalf of Paul HalesSent: Fri 8/11/2006 3:51 AMTo:
Asterisk Users Mailing
> > 2) Phone activation at check-in/phone de-activation and billing at
> > check-out. Are there GUI tools for this, or should I write my own
> > back/front end?
> >
The integration with the hotel systems for the activation/deactivation
and billing can be tricky. Check the archives for some discus
For the OP, do you have an entry against "Display Name" on the PSTN
tab, whilst logged in as admin/advanced? If I have an entry in this,
what you describe happens for me. If the field is empty, CLID is sent
correctly to my Asterisk box.
On 21/07/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
I ju
Anyone using Astersk in a way that connects the U.S. and China? I have a few questions that would be great to discuss offline.Thanks,Jonathan
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After trying a number of different solutions and having a very hard time to get non-technical people on the team to use them (over email, excel timesheets, etc), I have to recommendbasecamp from 37 signals.Its (so far) been very well used and adopted by the team.
jonathan
Very happy with the 501 and 601. So far, like the 430 as well.
The 301 is good for what it is but the display and lack of speakerphone
are annoying to me.
They are all very stable and compatible though. The provisioning on
these phones is excellent as well.
-Jonathan
-Original Message
Haven't read this whole thread (got way behind in this list :) )
Polycom has a softphone with video support also. Not sure if it is good
or not, just downloaded the trial version to test it out.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-user
ing around with using the redirect command to a
new context that plays the sound - this seems to work but there are
issues in then maintaining the connection.Any ideas?Thanks,Jonathan Palley
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gt; What kind of T1? TDM? Data? What type of signaling are you planning
> to use e&m? There is a lot of information that that question is
> lacking for anyone to advise you ...
>
> Jonathan Miller wrote:
> > I have a true leased line (a T1) between the two sites.
> &g
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Hash: SHA256
I have a true leased line (a T1) between the two sites.
What parts do I configure for Asterisk to utilized the link bi-directional?
On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote:
> On Wednesday 28 June 2006 08:48, Jonathan Mil
be set and how are they diffent than
just a standard PRI, which I have working now?
Thanks for your help,
Jonathan
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Version: PGP Universal 2.0.6
iQEVAwUBRKJtFpJhYmFK+jfsAQh/tggAiqCqlefhEyAuIcshX5AaMGx3flVdHn5C
mh1TY5i/Z8tf4LBEh+TuXvUFGNXvnPn12nrEwkF8s4HOUcDwV
at all, i will send tomorrow (i'm out of office today) the
unit back to Cortex Systems and i will put cleary on the box "faulty"
with some documents as the technical and sales consultant pointed me.
I've got an invoice and a UPS delivery note so no fear at all.
Thanks for all.
Be
Does the Sipura web interface on the info page reveal that the spa2100
is successfully receiving CLID?
My SPA2100 passes CLID from asterisk to the connected phone without problem.
On 23/06/06, Jim Lynch <[EMAIL PROTECTED]> wrote:
I have a Uniden wireless phone connected into Linksys/Supura 2100
I'll second that. I really like the provisioning features. My customers
prefer the 501 because they like the layout and speaker phone
functionality.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
West
Sent: Friday, June 23, 2006
t happening.
I changed the default RTP ports to be the same as those on the phones,
16384 > 32778
but am still having trouble getting the audio to play on this machine. I
suspect a package missing, but don't know what that could be and I'm
not able to find much help on this in the h
Could your register line require attention ? (2001?)
7960xxx:[EMAIL PROTECTED]/2001 - I thought your target was 2201?
On 13/06/06, Russell Horn <[EMAIL PROTECTED]> wrote:
Hi folks - I've recently returned to asterisk after an eighteen month break.
I've two sip providers - gradwell in the UK (i
have a dial tone?
On 6/12/06, Jonathan Attwood <[EMAIL PROTECTED]> wrote:
> Analogue Telephone Adapter(s)
> Linksys/Sipura range SPA-3102; SPA2100; SPA1001 to name but 3
>
> On 12/06/06, John Klimek <[EMAIL PROTECTED]> wrote:
> > Ok, I've done some more researc
Analogue Telephone Adapter(s)
Linksys/Sipura range SPA-3102; SPA2100; SPA1001 to name but 3
On 12/06/06, John Klimek <[EMAIL PROTECTED]> wrote:
Ok, I've done some more research and I don't think I want an FXO box...
What I'd like to do is use BroadVoice (with their BYOD plan) and then
run Aster
Thanks a lot for all your post... any other will be welcomed.
Kind regards,
Jonathan GF
On 5/25/06, Woodoo People .pGa! <[EMAIL PROTECTED]> wrote:
> i'm brand new and i would like to ask about soekris hardware. I read
> along the web but i have some doubts that i think c
es to traditional PBXs so echo isnt usually an
issue.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bart
Fisher
Sent: Tuesday, May 23, 2006 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] What
heard). It does not crash and all the other calls are unaffected.
Both systems are CentOS 4.2 fully updated and running Asterisk 1.2.7.
The peripheral system is Fedora Core 3 running Asterisk 1.2.7 also.
-Jonathan
___
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make a complete installation of
[EMAIL PROTECTED]
It seems the product is quite stable and i couldn't see this about
astlinux in any place. I would appreciate your thougts about astlinux
and some recomendations will be welcomed.
Thanks for you tu answer and for your magnificent document.
K
you select?
Any thoughts will be welcomed.
Kind regards,
Jonathan GF
On 5/17/06, olivier.taylor <[EMAIL PROTECTED]> wrote:
more kindly :
http://www.astlinux.org/
Olivier
Christopher Snell a écrit :
Google and voip-info.org will have answers to all of your questions.
On 5
ht over my
questions i would really love it.
Thanks in advance.
Kind regards,
Jonathan GF
On 5/17/06, Christopher Snell <[EMAIL PROTECTED]> wrote:
Google and voip-info.org will have answers to all of your questions.
On 5/17/06, Jonathan Gonzalez < [EMAIL PROTECTED]> wrote:
>
Hi gr
I believe I had to do the udev permissions file and also cause udevd to
launch at bootup before modprobe'ing zaptel stuff. Check to make sure
that udevd is launching automatically on bootup and that the udev rules
and permissions are in place.
-Jon
T.S wrote:
Yes I use Slackware 10.2, but I
INFO is the way to go for DTMF at least on the PSTN tab of your SPA3K
I have dtmfmode=auto in sip.conf & I use DISA daily
On 5/17/06, Philippe Lindheimer <[EMAIL PROTECTED]> wrote:
Just tried it on mine, worked fine:
Cellphone Call -> POTS -> SPA3000 -> Asterisk -> DISA -> Telasip
As an FYI,
f fax support into asterisk / [EMAIL PROTECTED] ?
5) It's posible to create personalized dialplans that enables a hidden
or passcode/password protected menu for remote administration or
remote use of the pbx?
Thanks in advance for your kind help and support.
Jonathan GF
--
si secretum tibi sit,
400mhz with 256MB
ram and a 9GB scsi hard drive.
Everything is working great even on such meager hardware.
Our other systems are Dual Xeon servers with 1 or 2GB of ram each
handling our PRI's and customer systems.
-Jonathan
___
--Bandwidt
done. It works but I still need to test it and
document it a little better. It is based on some scripts by other people
that I combined together.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matt
> Sent: Thurs
Wai Wu wrote:
> I notice those options. However, I was looking to start the recording
> through a third party control program. I know I can do this via
> chanspy, but is there better way?
Not that I know of... I was looking for something kind of similar, and
ended up actually using a conference, a
Peter Fern wrote:
> Probably because the Local proxy channel drops out once the two sides
> have been bridged. If you want the Local chan to stay up, use the /n
> parameter and the local channel won't perform the native transfer. This
> does have it's own problems, but should do what you want.
I searched through the archives and the wiki...don't be so pissy...i
missed it I guess, my bad....
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jerry Jones
> Sent: Wednesday, April 19, 2006 9:17 AM
>
eset=0) [ 0x9f, 0x8b, 0x01, 0x00,
0xa1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'Cell', 0x20,
'Phone', 0x20, 0x20, 0x20, 'KY' ]
-Jonathan
Jonathan Creasy
Network Engineer
BluegrassNet De
Try adding the following to sip.conf
--
[general]
progressinband=no
-Jon
Brent Torrenga wrote:
Anyone experience the "double ringing" when calling out over TelIAX? I am
using a Cisco 79[46]0, and do not use the "r" option in the Dial() command.
I always thought that the "r" is what
I could be wrong but off the top of my head I think that it is in the
features section of the config file.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Tuesday, April 18, 2006 4:47 PM
To: 'Asterisk Users Mailing List
> I can dial other extensions internally, and can get to voicemail, but
> when I try an outside number, I hear dial tone, the digits dialed, yet
> nothing happens when I press "Send".
>
> Nothing appears on the Asterisk CLI screen.
>
Did the upgrade modify the dialplan setting on your phone? Th
Does anyone know the format for the TOS element in the Polycom
config?
-Jonathan
Jonathan Creasy
Network Engineer
BluegrassNet Development
www.bgnd.com www.bluegrass.net
o. 502-589-4638
c. 502-889-5567
h. 502-541-0566
Anyone know what has happened to the local calling guide?
http://members.dandy.net/~czg/search.html
-Jonathan
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have a look here: http://nerdvittles.com/index.php?p=88
On 4/4/06, Wasif <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I need to receive FAX over DID and forward that FAX in email to particular
> person. I read some articles about www.voip-info.org but I am confused in
> HylaFax, IAXmodem & spandsp.
>
> C
ls using the set/check group commands.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup
Account codes are set either by using the Set function or the
accountcode= property in the SIP/IAX conf files.
-Jonathan
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.
B would never show the status of A and A would never show the proper
status of C. Phone C never showed the proper status of A or B.
I didn't spend any more time on it but I'll try and get a chance to day
to set the phones back up and give it a little more scientific testing.
Title: Re: [Asterisk-Users] Blocked channels,according to our telco... leading
to CONGESTION status
I’d say try it out and see what the
CPU load is. It’s not that hard to drop it in your dialplan and give it a
try. It’s much easier than figuring out all the possible variables in
your setup
I agree we have this working also.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Boris Bakchiev
> Sent: Friday, March 31, 2006 8:55 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] As
You have to use H323 the last time I did
anything with their equipment. It has been almost a year but I think it went
fairly smoothly. Do you have a specific question?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen Arulraj
Sent: Friday, March 31, 2006 5
I have found this to be true also.
[whatever] has to match username=
It appears that it ignores the username field for IAX users.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tomas Komarek
> Sent: Mond
's network and whenever we got a call from them that had one way
audio and we redirected that call in and then back out to another server
(on the weekends) the box would do this.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTE
cated networks built
for the phones and the Asterisk server acts as a dhcp, ntp and ftp
server as well as the PBX. The only devices on the networks were phones.
They use it as a really nice phone system and use "old fashioned"
termination.
-Jonathan
> -Original Message-
> F
ne or two phones.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nick
Hoffman
Sent: Saturday, March 25, 2006 3:06 AM
To: asterisk-users Mailing List
Subject: [Asterisk-Users] Polycom IP 301 is slow
Hi guys, I've been using a Polycom IP
Have you verified that ztdummy is loaded?
On Sun, 2006-03-26 at 01:06 -0500, Erick Perez wrote:
> Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel) no
> hardware interfaces installed gives me this error. Im a bit new to
> this so any help will be appreciated.
>
> == Parsing '/et
I must be missing something here. Have you tried option "g" on your
dial command to the acd server? If option g is not specified, then dial
will hangup the call when exiting regaurdless of what the other iax box
did.
-Jon
Douglas Garstang wrote:
I just changed the macro to:
exten => s,
Hi, i'm a newbie running Asterisk 1.2.1 with Cisco 7940/7960 SIP version 7.4 phones. Is
there any way in the dial plan or other mystical conf file to allow a user whose
extension is presently ringing to press a button on their phone that would instantly send
the incoming call to the called use
I have had very reliable inbound/outbound service from Junction Networks
(www.junctionnetworks.com). The one time I did have an issue, it was
resolved quickly. During my testing I concluded that BroadVoice (my
partner refers to them as NoVoice) was unreliable (approximately 40% of
all of our test
It's a toll free number. You can call it from anywhere and the costs of the
call go on the callee not the caller.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of C F
> Sent: Wednesday, March 22, 2006 7:50 AM
> To: Asterisk Users Mai
Do you want to dial an outgoing line as well as the SIP line?
Dial(SIP/&${OUTGOING}/${EXTEN}) ?
I can't say obviously without more info but it sounds to me like you are
looking for the wrong solution
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [
We are doing this with the latest spandsp, iaxmodem and hylafax.
Seems to work very well for us so far.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Gaudette
> Sent: Tuesday, March 21, 2006
I am having this problem also. I have 2 systems running 1.2.5. I had the
problem and one system was running 1.2.4 and the other was running a CVS
HEAD from October so I upgraded them both to 1.2.5 with no success.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:as
l Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
> Augenstine
> Sent: March 17, 2006 5:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Question about meetme app
>
> A locked conferen
A locked conference means that a pin number is required to join the
conference.
On Fri, 2006-03-17 at 16:20 -0500, Michael Gaudette wrote:
> I have a quick question about the MeetMe app. A locked conference means
> what exactly?
>
> A) That people can't join anymore
> B) That everyone is muted
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
given that AAH is asterisk + amp + bunch of other stuff, can't you just
setup incoming routing and outgoing routing within AMP?
Ira wrote:
> At 09:04 PM 03/16/2006, you wrote:
>> 6 different companies with 6 different IVR's and different ring groups
>
Jeff Hoppe wrote:
When you say that you tweaked the volumes, is that modifying the
Asterisk code to call sox and soxmix or are you mixing outside of
Asterisk. Also, I used Sox to increase the volume and then Soxmix to
mix the two audio files. Is there a way to just use soxmix to
increase the vol
Well, whether I SHOULD get it or not may be totally irrelevant to
whether I CAN or DO get it. The caller ID info is most definitely there
and it shows up in my CDR records. However, it is not displayed on the
device because only the number is allowed on our PRI.
-Jonathan
-Original Message
Well for one thing, on a PRI it is usually still transmitted with a bit
set that tells the system to hide it.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Thursday, March 16, 2006 9:49 PM
To: Asterisk Users Mailing
nce a week and since upgrading to 1.2.5 this
morning from 1.2.4 it has happened twice.
Does anyone know what is happening here?
-Jonathan
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On 3/10/06, Sharath Chandra <[EMAIL PROTECTED]> wrote:
>
> How can i configure the following scenario,
>
> - User 'A' dials into Asterisk,
> - Asterisk puts user 'A' on hold
> - Dials Out to User 'B'
> - Consults user B' if he wants to take the call (Press 1)
Try this:
musiconhold.conf:
[stream2]
mode=mp3
directory=http://pubint.ic.llnwd.net/stream/pubint_wnpr
extensions.conf:
exten => 1234,1,Answer
exten => 1234,2,MusicOnHold(stream2)
exten => 1234,3,Hangup
On Wed, 2006-02-22 at 09:28 -0700, Douglas Garstang wrote:
> Ok, I'm tearing my hair out
Barix Instreamer takes RCA in and MP3 or ulaw stream out. Asterisk can
use either for MOH.
On Fri, 2006-02-17 at 07:42 -0600, Rich Adamson wrote:
> Been around asterisk for two-plus years, but need a little input from the
> list on this topic.
>
> Have a potential client that wants to replace th
Upgrade to 1.2.4 - bug in 1.2.2 - see www.asterisk.org front page.
-Jon
Kamran Ahmad wrote:
Hi all
I am using Asterisk 1.2.2 on frdora core 4. i have two
sip UA. if i put canreinvite=yes voice Ok on both
sides. and if i change canreinvite=no there is no
voice (media through asterisk)
one t
-09 at 14:09 -0800, Jonathan Feally wrote:
Hello All,
I'm looking to get some feedback on which solution of providing FXS is
going to have the best results with faxing. I'm only looking to see what
method is going to provide the best digitization into Asterisk, not for
tr
Hello All,
I'm looking to get some feedback on which solution of providing FXS is
going to have the best results with faxing. I'm only looking to see what
method is going to provide the best digitization into Asterisk, not for
transmission from Asterisk to else where. Any recommendations of
s
This feature also works on the IP301 phones. The obvious caveat is that
it is one-way only. Still nice for an "all-page" though.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Se
BOFH told me he uses it to listen to his co-workers
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
> Sent: Thursday, February 09, 2006 12:27 PM
> To: asterisk-users@lists.digium.com
> Subject: SOLVED: Re: [Asterisk-U
From me looking at it - it looks like the Telco is not accepting a
3 digit number. Have you tried 411 on the PRI to see if you are getting
the same error?
My 2 Cents
-Jon
Michael Collins wrote:
Joe,
It is
entirely possible, even probable,
that you spoke with someone
Do people not use the Grandstream ATA's because they are cheap or
because there is actually a problem with them?
They have a 2 line version for around $50 that I have used in various
locations. I have about 8 or so. They seem to do an excellent job.
-Jonathan
-Original Message-
It's something like exten => 15,1,Dial(Console/DSP)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Azzopardi
Sent: Saturday, February 04, 2006 2:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How ca
I have been planning to do the same thing but never got around to it, I
actually did write a nice class to wrap the interface to the manager but
it isn't complete.
Would you be willing to share your work?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROT
The Grandstream ATA (480 I think...) does this and usually costs less
than the Sipura. It has 1 FXS and 1 FXO.
-Jonathan
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Damon Estep
> Sent: Thursday, February 02, 2006
Anyone in Winnipeg Canada?
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x with "astgenkey -n
office.pbx.bluegrass.net" using the host name for each box of course.
I then copied the .pub files to the /var/lib/asterisk/keys folder from
each box to the other box.
What am I missing?
-Jonathan
___
--Bandwi
On 1/24/06, Peter Bowyer <[EMAIL PROTECTED]> wrote:
> On 24/01/06, scott <[EMAIL PROTECTED]> wrote:
> > Hi
> >
> > Does anyone know a UK Voip Proivder that will give me more than 1 telephone
> > number and point it to my sip account.
> >
> > www.SipGate.co.uk are great but they only allow 1 teleph
You can try Voxboned(www.voxbone.com) if you need inbound only.
On Tue, 2006-01-24 at 09:13 +, scott wrote:
> Hi
>
> Does anyone know a UK Voip Proivder that will give me more than 1 telephone
> number and point it to my sip account.
>
> www.SipGate.co.uk are great but they only allow 1 te
This basicaly means you need to recompile the kernel with HZ=1000.
On a 2.6.x kernel in make menuconfig you can find this under
Processor type and features --->
Timer frequency (1000 HZ) --->
100,250,1000
-Jon
Sean Cook wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Been run
ugh that nat=yes did not effect this. And if it does why does it
native bridge ok on inbound calls with the same nat=yes
On 1/15/06, Jonathan Feally <[EMAIL PROTECTED]>
wrote:
I'm guessing that you have a similar entry in your sip.conf for the
7960?? The 7960 has a setting
I'm guessing that you have a similar entry in your sip.conf for the
7960?? The 7960 has a setting for preferred codec. It defaults to g711
U-Law. You might try changing this setting also as the 7960 doesn't
know that you only want to speak A-Law. You will also want to make sure
that the nat set
You need the unit manager software that should have come with your box.
Your box most likely only speaks SNMP, so this is the only tool I know
that has the MIB's and setup to know how to set the MIB values. However
there are many more tweaks in manually tuning some of the MIB's through
the unit
I've been looking around to see where I could get one, would solve a
problem we have quite nicely.
Where did you get yours if I may ask?
-Jonathan
Jonathan O'Connor
Senior System Administrator
Inoveris LLC
Direct Line (614) 791-3742
Fax (614) 791-3748
Helpdesk 86
getis very quiet.Thanks for your helpRegardsJonathan On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s Asenjo Gonz?lez wrote:
> Hi everybody!
>
> Jonathan wrote:
> >
> > Hi,
> >
> > I'm using a td400p card with an FXO port and asterisk 1.2.1 in South
> > Korea
Hi,
I'm struggling to
get busydetect to work.
I'm using asterisk
1.2.1 and a digium TDM04B (4 port FXO) card.
I've set
busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've
modified zondata.c with a busy setting of 620+480, 300/200 which is the
busysignal receive
It certainly does.
How many rules can you create in the port forwarding section of the V2900?
I was told that the V2900 has SIP_ALG. Is this something you've activated?
On 1/7/06, Faris Raouf <[EMAIL PROTECTED]> wrote:
> Jonathan Attwood wrote:
> > I'm in conversation
I'm in conversation with Draytek's pre-sales dept..
Here's the most recent reply:
All I want to know is, if I buy one of these routers, will it break my setup
or not - ie. assuming I set up the relevant port-forwarding, can I
expect any one-way audio issues. Can't get a definitive a
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